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gir-files: Update to gstreamer master
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@ -294,7 +294,7 @@ filled.</doc>
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<doc xml:space="preserve">Check if appsink will emit the "new-preroll" and "new-sample" signals.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">%TRUE if @appsink is emiting the "new-preroll" and "new-sample"
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<doc xml:space="preserve">%TRUE if @appsink is emitting the "new-preroll" and "new-sample"
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signals.</doc>
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<type name="gboolean" c:type="gboolean"/>
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</return-value>
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@ -1032,7 +1032,7 @@ mode when implementing various network protocols or hardware devices.
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The pull mode, in which the need-data signal triggers the next push-buffer call.
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This mode is typically used in the "random-access" stream-type. Use this
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mode for file access or other randomly accessable sources. In this mode, a
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mode for file access or other randomly accessible sources. In this mode, a
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buffer of exactly the amount of bytes given by the need-data signal should be
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pushed into appsrc.
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@ -1052,7 +1052,7 @@ occurs or the state of the appsrc has gone through READY.</doc>
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element is the last buffer of the stream.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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@ -1096,9 +1096,9 @@ When the block property is TRUE, this function can block until free
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space becomes available in the queue.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -1121,9 +1121,9 @@ When the block property is TRUE, this function can block until free
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space becomes available in the queue.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -1150,9 +1150,9 @@ When the block property is TRUE, this function can block until free
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space becomes available in the queue.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -1186,7 +1186,7 @@ extracted</doc>
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element is the last buffer of the stream.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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@ -1328,9 +1328,9 @@ When the block property is TRUE, this function can block until free
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space becomes available in the queue.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -1353,9 +1353,9 @@ When the block property is TRUE, this function can block until free
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space becomes available in the queue.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -1382,9 +1382,9 @@ When the block property is TRUE, this function can block until free
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space becomes available in the queue.</doc>
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -1843,9 +1843,9 @@ gst_app_src_set_callbacks().</doc>
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<callback name="push_buffer">
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -1864,7 +1864,7 @@ gst_app_src_set_callbacks().</doc>
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<callback name="end_of_stream">
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the EOS was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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@ -1880,9 +1880,9 @@ gst_app_src_set_callbacks().</doc>
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<callback name="push_sample">
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -1902,9 +1902,9 @@ extracted</doc>
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<callback name="push_buffer_list">
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfuly queued.
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<doc xml:space="preserve">#GST_FLOW_OK when the buffer list was successfully queued.
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#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
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#GST_FLOW_EOS when EOS occured.</doc>
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#GST_FLOW_EOS when EOS occurred.</doc>
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<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
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</return-value>
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<parameters>
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@ -324,14 +324,14 @@ and/or use gtk-doc annotations. -->
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</parameters>
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</function-macro>
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<constant name="AUDIO_CONVERTER_OPT_DITHER_METHOD" value="GstAudioConverter.dither-method" c:type="GST_AUDIO_CONVERTER_OPT_DITHER_METHOD">
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<doc xml:space="preserve">#GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
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<doc xml:space="preserve">#GstAudioDitherMethod, The dither method to use when
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changing bit depth.
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Default is #GST_AUDIO_DITHER_NONE.</doc>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_CONVERTER_OPT_MIX_MATRIX" value="GstAudioConverter.mix-matrix" c:type="GST_AUDIO_CONVERTER_OPT_MIX_MATRIX">
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<doc xml:space="preserve">#GST_TYPE_VALUE_LIST, The channel mapping matrix.
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<doc xml:space="preserve">#GST_TYPE_LIST, The channel mapping matrix.
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The matrix coefficients must be between -1 and 1: the number of rows is equal
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to the number of output channels and the number of columns is equal to the
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@ -362,7 +362,7 @@ g_value_unset (&v);
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD" value="GstAudioConverter.noise-shaping-method" c:type="GST_AUDIO_CONVERTER_OPT_NOISE_SHAPING_METHOD">
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<doc xml:space="preserve">#GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use
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<doc xml:space="preserve">#GstAudioNoiseShapingMethod, The noise shaping method to use
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to mask noise from quantization errors.
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Default is #GST_AUDIO_NOISE_SHAPING_NONE.</doc>
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@ -376,7 +376,7 @@ Default is 1</doc>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_CONVERTER_OPT_RESAMPLER_METHOD" value="GstAudioConverter.resampler-method" c:type="GST_AUDIO_CONVERTER_OPT_RESAMPLER_METHOD">
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<doc xml:space="preserve">#GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when
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<doc xml:space="preserve">#GstAudioResamplerMethod, The resampler method to use when
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changing sample rates.
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Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.</doc>
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@ -957,7 +957,7 @@ See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values</doc>
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<type name="utf8" c:type="gchar*"/>
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</constant>
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<constant name="AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION" value="GstAudioResampler.filter-interpolation" c:type="GST_AUDIO_RESAMPLER_OPT_FILTER_INTERPOLATION">
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<doc xml:space="preserve">GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coeficients should be
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<doc xml:space="preserve">GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coefficients should be
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interpolated.
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GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.</doc>
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@ -1219,6 +1219,10 @@ downstream specifies a range or a set of acceptable rates).</doc>
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<property name="output-buffer-duration" writable="1" transfer-ownership="none">
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<type name="guint64" c:type="guint64"/>
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</property>
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<property name="output-buffer-duration-fraction" version="1.18" writable="1" transfer-ownership="none">
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<doc xml:space="preserve">Output block size in nanoseconds, expressed as a fraction.</doc>
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<type name="Gst.Fraction"/>
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</property>
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<field name="parent">
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<type name="GstBase.Aggregator" c:type="GstAggregator"/>
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</field>
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@ -2733,7 +2737,7 @@ This is expressed in caps by having a single channel and no channel mask.
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This is expressed in caps by having a channel mask with no bits set.
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As another special case it is allowed to have two channels without a channel mask.
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This implicitely means that this is a stereo stream with a front left and front right
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This implicitly means that this is a stereo stream with a front left and front right
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channel.</doc>
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<member name="none" value="-3" c:identifier="GST_AUDIO_CHANNEL_POSITION_NONE" glib:nick="none">
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<doc xml:space="preserve">used for position-less channels, e.g.
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@ -3043,7 +3047,7 @@ The object can perform conversion of:
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<doc xml:space="preserve">Create a new #GstAudioConverter that is able to convert between @in and @out
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audio formats.
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@config contains extra configuration options, see #GST_AUDIO_CONVERTER_OPT_*
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@config contains extra configuration options, see `GST_AUDIO_CONVERTER_OPT_*`
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parameters for details about the options and values.</doc>
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<return-value transfer-ownership="full">
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@ -3306,7 +3310,7 @@ If the parameters in @config can not be set exactly, this function returns
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%FALSE and will try to update as much state as possible. The new state can
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then be retrieved and refined with gst_audio_converter_get_config().
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Look at the #GST_AUDIO_CONVERTER_OPT_* fields to check valid configuration
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Look at the `GST_AUDIO_CONVERTER_OPT_*` fields to check valid configuration
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option and values.</doc>
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<return-value transfer-ownership="none">
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@ -3732,7 +3736,7 @@ invalidated by a call to this function.</doc>
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<doc xml:space="preserve">a #GstAudioDecoder</doc>
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<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
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</instance-parameter>
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<parameter name="buf" transfer-ownership="none">
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<parameter name="buf" transfer-ownership="full" nullable="1" allow-none="1">
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<doc xml:space="preserve">decoded data</doc>
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<type name="Gst.Buffer" c:type="GstBuffer*"/>
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</parameter>
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@ -3766,7 +3770,7 @@ invalidated by a call to this function.</doc>
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<doc xml:space="preserve">a #GstAudioDecoder</doc>
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<type name="AudioDecoder" c:type="GstAudioDecoder*"/>
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</instance-parameter>
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<parameter name="buf" transfer-ownership="none">
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<parameter name="buf" transfer-ownership="full" nullable="1" allow-none="1">
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<doc xml:space="preserve">decoded data</doc>
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<type name="Gst.Buffer" c:type="GstBuffer*"/>
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</parameter>
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@ -5075,7 +5079,7 @@ may be invalidated by a call to this function.</doc>
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<doc xml:space="preserve">a #GstAudioEncoder</doc>
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<type name="AudioEncoder" c:type="GstAudioEncoder*"/>
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</instance-parameter>
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<parameter name="buffer" transfer-ownership="none">
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<parameter name="buffer" transfer-ownership="full" nullable="1" allow-none="1">
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<doc xml:space="preserve">encoded data</doc>
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<type name="Gst.Buffer" c:type="GstBuffer*"/>
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</parameter>
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@ -6724,7 +6728,7 @@ meta as well as extracting it.</doc>
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<member name="truncate_range" value="1" c:identifier="GST_AUDIO_PACK_FLAG_TRUNCATE_RANGE" glib:nick="truncate-range">
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<doc xml:space="preserve">When the source has a smaller depth
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than the target format, set the least significant bits of the target
|
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to 0. This is likely sightly faster but less accurate. When this flag
|
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to 0. This is likely slightly faster but less accurate. When this flag
|
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is not specified, the most significant bits of the source are duplicated
|
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in the least significant bits of the destination.</doc>
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</member>
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|
@ -6900,7 +6904,7 @@ frames.</doc>
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@in_frames are given to @resampler.</doc>
|
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<return-value transfer-ownership="none">
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<doc xml:space="preserve">The number of frames that would be availabe after giving
|
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<doc xml:space="preserve">The number of frames that would be available after giving
|
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@in_frames as input to @resampler.</doc>
|
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<type name="gsize" c:type="gsize"/>
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</return-value>
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@ -7080,11 +7084,11 @@ for @quality in @options.</doc>
|
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</member>
|
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<member name="linear" value="1" c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_LINEAR" glib:nick="linear">
|
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<doc xml:space="preserve">linear interpolation of the
|
||||
filter coeficients.</doc>
|
||||
filter coefficients.</doc>
|
||||
</member>
|
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<member name="cubic" value="2" c:identifier="GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC" glib:nick="cubic">
|
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<doc xml:space="preserve">cubic interpolation of the
|
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filter coeficients.</doc>
|
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filter coefficients.</doc>
|
||||
</member>
|
||||
</enumeration>
|
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<enumeration name="AudioResamplerFilterMode" glib:type-name="GstAudioResamplerFilterMode" glib:get-type="gst_audio_resampler_filter_mode_get_type" c:type="GstAudioResamplerFilterMode">
|
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|
@ -7242,7 +7246,7 @@ FALSE on error.</doc>
|
|||
</parameters>
|
||||
</virtual-method>
|
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<virtual-method name="clear_all" invoker="clear_all">
|
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<doc xml:space="preserve">Fill the ringbuffer with silence.
|
||||
<doc xml:space="preserve">Clear all samples from the ringbuffer.
|
||||
|
||||
MT safe.</doc>
|
||||
|
||||
|
@ -7335,11 +7339,11 @@ usually less than the segment size but can be bigger when the
|
|||
implementation uses another internal buffer between the audio
|
||||
device.
|
||||
|
||||
For playback ringbuffers this is the amount of samples transfered from the
|
||||
For playback ringbuffers this is the amount of samples transferred from the
|
||||
ringbuffer to the device but still not played.
|
||||
|
||||
For capture ringbuffers this is the amount of samples in the device that are
|
||||
not yet transfered to the ringbuffer.</doc>
|
||||
not yet transferred to the ringbuffer.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
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<doc xml:space="preserve">The number of samples queued in the audio device.
|
||||
|
@ -7531,7 +7535,7 @@ MT safe.</doc>
|
|||
</parameters>
|
||||
</method>
|
||||
<method name="clear_all" c:identifier="gst_audio_ring_buffer_clear_all">
|
||||
<doc xml:space="preserve">Fill the ringbuffer with silence.
|
||||
<doc xml:space="preserve">Clear all samples from the ringbuffer.
|
||||
|
||||
MT safe.</doc>
|
||||
|
||||
|
@ -7655,11 +7659,11 @@ usually less than the segment size but can be bigger when the
|
|||
implementation uses another internal buffer between the audio
|
||||
device.
|
||||
|
||||
For playback ringbuffers this is the amount of samples transfered from the
|
||||
For playback ringbuffers this is the amount of samples transferred from the
|
||||
ringbuffer to the device but still not played.
|
||||
|
||||
For capture ringbuffers this is the amount of samples in the device that are
|
||||
not yet transfered to the ringbuffer.</doc>
|
||||
not yet transferred to the ringbuffer.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">The number of samples queued in the audio device.
|
||||
|
@ -8548,6 +8552,17 @@ All scheduling of samples and timestamps is done in this base class
|
|||
together with #GstAudioBaseSink using a default implementation of a
|
||||
#GstAudioRingBuffer that uses threads.</doc>
|
||||
|
||||
<virtual-method name="clear_all">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="sink" transfer-ownership="none">
|
||||
<type name="AudioSink" c:type="GstAudioSink*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="close">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -8581,6 +8596,17 @@ together with #GstAudioBaseSink using a default implementation of a
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="pause">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="sink" transfer-ownership="none">
|
||||
<type name="AudioSink" c:type="GstAudioSink*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="prepare">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -8606,6 +8632,28 @@ together with #GstAudioBaseSink using a default implementation of a
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="resume">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="sink" transfer-ownership="none">
|
||||
<type name="AudioSink" c:type="GstAudioSink*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="stop">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="sink" transfer-ownership="none">
|
||||
<type name="AudioSink" c:type="GstAudioSink*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="unprepare">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -8647,7 +8695,6 @@ together with #GstAudioBaseSink using a default implementation of a
|
|||
</field>
|
||||
</class>
|
||||
<record name="AudioSinkClass" c:type="GstAudioSinkClass" glib:is-gtype-struct-for="AudioSink">
|
||||
<doc xml:space="preserve">#GstAudioSink class. Override the vmethods to implement functionality.</doc>
|
||||
|
||||
<field name="parent_class">
|
||||
<doc xml:space="preserve">the parent class structure.</doc>
|
||||
|
@ -8753,8 +8800,60 @@ together with #GstAudioBaseSink using a default implementation of a
|
|||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="pause">
|
||||
<callback name="pause">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="sink" transfer-ownership="none">
|
||||
<type name="AudioSink" c:type="GstAudioSink*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="resume">
|
||||
<callback name="resume">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="sink" transfer-ownership="none">
|
||||
<type name="AudioSink" c:type="GstAudioSink*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="stop">
|
||||
<callback name="stop">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="sink" transfer-ownership="none">
|
||||
<type name="AudioSink" c:type="GstAudioSink*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="clear_all">
|
||||
<callback name="clear_all">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="sink" transfer-ownership="none">
|
||||
<type name="AudioSink" c:type="GstAudioSink*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="_gst_reserved" readable="0" private="1">
|
||||
<array zero-terminated="0" fixed-size="4">
|
||||
<array zero-terminated="0" fixed-size="0">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</array>
|
||||
</field>
|
||||
|
@ -9235,7 +9334,7 @@ of the current one.</doc>
|
|||
</method>
|
||||
<method name="set_rate" c:identifier="gst_audio_stream_align_set_rate" version="1.14">
|
||||
<doc xml:space="preserve">Sets @rate as new sample rate for the following processing. If the sample
|
||||
rate differs this implicitely marks the next data as discontinuous.</doc>
|
||||
rate differs this implicitly marks the next data as discontinuous.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
|
@ -9458,13 +9557,6 @@ rate differs this implicitely marks the next data as discontinuous.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</function-macro>
|
||||
<function-macro name="IS_STREAM_VOLUME" c:identifier="GST_IS_STREAM_VOLUME" introspectable="0">
|
||||
|
||||
<parameters>
|
||||
<parameter name="obj">
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function-macro>
|
||||
<constant name="META_TAG_AUDIO_CHANNELS_STR" value="channels" c:type="GST_META_TAG_AUDIO_CHANNELS_STR" version="1.2">
|
||||
<doc xml:space="preserve">This metadata stays relevant as long as channels are unchanged.</doc>
|
||||
|
||||
|
@ -9480,17 +9572,10 @@ rate differs this implicitely marks the next data as discontinuous.</doc>
|
|||
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</constant>
|
||||
<function-macro name="STREAM_VOLUME" c:identifier="GST_STREAM_VOLUME" introspectable="0">
|
||||
|
||||
<parameters>
|
||||
<parameter name="obj">
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function-macro>
|
||||
<function-macro name="STREAM_VOLUME_GET_INTERFACE" c:identifier="GST_STREAM_VOLUME_GET_INTERFACE" introspectable="0">
|
||||
|
||||
<parameters>
|
||||
<parameter name="inst">
|
||||
<parameter name="obj">
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function-macro>
|
||||
|
|
|
@ -180,8 +180,8 @@ gst_adapter_offset_at_discont(). The number of bytes that were consumed
|
|||
since then can be queried with gst_adapter_distance_from_discont().
|
||||
|
||||
A last thing to note is that while #GstAdapter is pretty optimized,
|
||||
merging buffers still might be an operation that requires a malloc() and
|
||||
memcpy() operation, and these operations are not the fastest. Because of
|
||||
merging buffers still might be an operation that requires a `malloc()` and
|
||||
`memcpy()` operation, and these operations are not the fastest. Because of
|
||||
this, some functions like gst_adapter_available_fast() are provided to help
|
||||
speed up such cases should you want to. To avoid repeated memory allocations,
|
||||
gst_adapter_copy() can be used to copy data into a (statically allocated)
|
||||
|
@ -1120,6 +1120,22 @@ sent before pushing the buffer.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="negotiate" invoker="negotiate" version="1.18">
|
||||
<doc xml:space="preserve">Negotiates src pad caps with downstream elements.
|
||||
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
|
||||
if #GstAggregatorClass.negotiate() fails.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="self" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstAggregator</doc>
|
||||
<type name="Aggregator" c:type="GstAggregator*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="negotiated_src_caps">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -1171,6 +1187,23 @@ sent before pushing the buffer.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="sink_event_pre_queue">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="aggregator" transfer-ownership="none">
|
||||
<type name="Aggregator" c:type="GstAggregator*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="aggregator_pad" transfer-ownership="none">
|
||||
<type name="AggregatorPad" c:type="GstAggregatorPad*"/>
|
||||
</parameter>
|
||||
<parameter name="event" transfer-ownership="none">
|
||||
<type name="Gst.Event" c:type="GstEvent*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="sink_query">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -1188,6 +1221,23 @@ sent before pushing the buffer.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="sink_query_pre_queue">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="aggregator" transfer-ownership="none">
|
||||
<type name="Aggregator" c:type="GstAggregator*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="aggregator_pad" transfer-ownership="none">
|
||||
<type name="AggregatorPad" c:type="GstAggregatorPad*"/>
|
||||
</parameter>
|
||||
<parameter name="query" transfer-ownership="none">
|
||||
<type name="Gst.Query" c:type="GstQuery*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="src_activate">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -1349,6 +1399,22 @@ Typically only called by subclasses.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="negotiate" c:identifier="gst_aggregator_negotiate" version="1.18">
|
||||
<doc xml:space="preserve">Negotiates src pad caps with downstream elements.
|
||||
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
|
||||
if #GstAggregatorClass.negotiate() fails.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="self" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstAggregator</doc>
|
||||
<type name="Aggregator" c:type="GstAggregator*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_latency" c:identifier="gst_aggregator_set_latency">
|
||||
<doc xml:space="preserve">Lets #GstAggregator sub-classes tell the baseclass what their internal
|
||||
latency is. Will also post a LATENCY message on the bus so the pipeline
|
||||
|
@ -1390,7 +1456,7 @@ can reconfigure its global latency.</doc>
|
|||
</parameters>
|
||||
</method>
|
||||
<method name="simple_get_next_time" c:identifier="gst_aggregator_simple_get_next_time" version="1.16">
|
||||
<doc xml:space="preserve">This is a simple #GstAggregator::get_next_time implementation that
|
||||
<doc xml:space="preserve">This is a simple #GstAggregatorClass.get_next_time() implementation that
|
||||
just looks at the #GstSegment on the srcpad of the aggregator and bases
|
||||
the next time on the running time there.
|
||||
|
||||
|
@ -1408,6 +1474,23 @@ and you have a dead line based aggregator subclass.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="update_segment" c:identifier="gst_aggregator_update_segment" version="1.18">
|
||||
<doc xml:space="preserve">Subclasses should use this to update the segment on their
|
||||
source pad, instead of directly pushing new segment events
|
||||
downstream.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="self" transfer-ownership="none">
|
||||
<type name="Aggregator" c:type="GstAggregator*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="segment" transfer-ownership="none">
|
||||
<type name="Gst.Segment" c:type="GstSegment*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<property name="latency" writable="1" transfer-ownership="none">
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</property>
|
||||
|
@ -1760,14 +1843,67 @@ _finish_buffer from inside that function.</doc>
|
|||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="negotiate">
|
||||
<callback name="negotiate">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="self" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstAggregator</doc>
|
||||
<type name="Aggregator" c:type="GstAggregator*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="sink_event_pre_queue">
|
||||
<callback name="sink_event_pre_queue">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="Gst.FlowReturn" c:type="GstFlowReturn"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="aggregator" transfer-ownership="none">
|
||||
<type name="Aggregator" c:type="GstAggregator*"/>
|
||||
</parameter>
|
||||
<parameter name="aggregator_pad" transfer-ownership="none">
|
||||
<type name="AggregatorPad" c:type="GstAggregatorPad*"/>
|
||||
</parameter>
|
||||
<parameter name="event" transfer-ownership="none">
|
||||
<type name="Gst.Event" c:type="GstEvent*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="sink_query_pre_queue">
|
||||
<callback name="sink_query_pre_queue">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="aggregator" transfer-ownership="none">
|
||||
<type name="Aggregator" c:type="GstAggregator*"/>
|
||||
</parameter>
|
||||
<parameter name="aggregator_pad" transfer-ownership="none">
|
||||
<type name="AggregatorPad" c:type="GstAggregatorPad*"/>
|
||||
</parameter>
|
||||
<parameter name="query" transfer-ownership="none">
|
||||
<type name="Gst.Query" c:type="GstQuery*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="_gst_reserved" readable="0" private="1">
|
||||
<array zero-terminated="0" fixed-size="20">
|
||||
<array zero-terminated="0" fixed-size="17">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</array>
|
||||
</field>
|
||||
</record>
|
||||
<class name="AggregatorPad" c:symbol-prefix="aggregator_pad" c:type="GstAggregatorPad" version="1.14" parent="Gst.Pad" glib:type-name="GstAggregatorPad" glib:get-type="gst_aggregator_pad_get_type" glib:type-struct="AggregatorPadClass">
|
||||
<doc xml:space="preserve">Pads managed by a #GstAggregor subclass.
|
||||
<doc xml:space="preserve">Pads managed by a #GstAggregator subclass.
|
||||
|
||||
This class used to live in gst-plugins-bad and was moved to core.</doc>
|
||||
|
||||
|
@ -2396,7 +2532,7 @@ a parser and share a lot of rather complex code.
|
|||
|
||||
* During the parsing process #GstBaseParseClass will handle both srcpad
|
||||
and sinkpad events. They will be passed to subclass if
|
||||
#GstBaseParseClass.event() or #GstBaseParseClass.src_event()
|
||||
#GstBaseParseClass.sink_event() or #GstBaseParseClass.src_event()
|
||||
implementations have been provided.
|
||||
|
||||
## Shutdown phase
|
||||
|
@ -3950,6 +4086,25 @@ information about the render delay.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_stats" c:identifier="gst_base_sink_get_stats" version="1.18">
|
||||
<doc xml:space="preserve">Return various #GstBaseSink statistics. This function returns a #GstStructure
|
||||
with name `application/x-gst-base-sink-stats` with the following fields:
|
||||
|
||||
- "average-rate" G_TYPE_DOUBLE average frame rate
|
||||
- "dropped" G_TYPE_UINT64 Number of dropped frames
|
||||
- "rendered" G_TYPE_UINT64 Number of rendered frames</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">pointer to #GstStructure</doc>
|
||||
<type name="Gst.Structure" c:type="GstStructure*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="sink" transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GstBaseSink</doc>
|
||||
<type name="BaseSink" c:type="GstBaseSink*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_sync" c:identifier="gst_base_sink_get_sync">
|
||||
<doc xml:space="preserve">Checks if @sink is currently configured to synchronize against the
|
||||
clock.</doc>
|
||||
|
@ -4460,6 +4615,15 @@ media. This property will add additional latency to the device in order to
|
|||
make other sinks compensate for the delay.</doc>
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</property>
|
||||
<property name="stats" version="1.18" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Various #GstBaseSink statistics. This property returns a #GstStructure
|
||||
with name `application/x-gst-base-sink-stats` with the following fields:
|
||||
|
||||
- "average-rate" G_TYPE_DOUBLE average frame rate
|
||||
- "dropped" G_TYPE_UINT64 Number of dropped frames
|
||||
- "rendered" G_TYPE_UINT64 Number of rendered frames</doc>
|
||||
<type name="Gst.Structure"/>
|
||||
</property>
|
||||
<property name="sync" writable="1" transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</property>
|
||||
|
@ -5074,6 +5238,7 @@ implementation will call alloc and fill.</doc>
|
|||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="get_caps">
|
||||
<doc xml:space="preserve">Called to get the caps to report.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="Gst.Caps" c:type="GstCaps*"/>
|
||||
|
@ -5082,7 +5247,7 @@ implementation will call alloc and fill.</doc>
|
|||
<instance-parameter name="src" transfer-ownership="none">
|
||||
<type name="BaseSrc" c:type="GstBaseSrc*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="filter" transfer-ownership="none">
|
||||
<parameter name="filter" transfer-ownership="none" nullable="1" allow-none="1">
|
||||
<type name="Gst.Caps" c:type="GstCaps*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
|
@ -5134,13 +5299,22 @@ out. The base class will sync on the clock using these times.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="negotiate">
|
||||
<virtual-method name="negotiate" invoker="negotiate" version="1.18">
|
||||
<doc xml:space="preserve">Negotiates src pad caps with downstream elements.
|
||||
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
|
||||
if #GstBaseSrcClass.negotiate() fails.
|
||||
|
||||
Do not call this in the #GstBaseSrcClass.fill() vmethod. Call this in
|
||||
#GstBaseSrcClass.create() or in #GstBaseSrcClass.alloc(), _before_ any
|
||||
buffer is allocated.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="src" transfer-ownership="none">
|
||||
<doc xml:space="preserve">base source instance</doc>
|
||||
<type name="BaseSrc" c:type="GstBaseSrc*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
|
@ -5334,9 +5508,29 @@ by the src; unref it after usage.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="negotiate" c:identifier="gst_base_src_negotiate" version="1.18">
|
||||
<doc xml:space="preserve">Negotiates src pad caps with downstream elements.
|
||||
Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
|
||||
if #GstBaseSrcClass.negotiate() fails.
|
||||
|
||||
Do not call this in the #GstBaseSrcClass.fill() vmethod. Call this in
|
||||
#GstBaseSrcClass.create() or in #GstBaseSrcClass.alloc(), _before_ any
|
||||
buffer is allocated.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="src" transfer-ownership="none">
|
||||
<doc xml:space="preserve">base source instance</doc>
|
||||
<type name="BaseSrc" c:type="GstBaseSrc*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="new_seamless_segment" c:identifier="gst_base_src_new_seamless_segment">
|
||||
<doc xml:space="preserve">Prepare a new seamless segment for emission downstream. This function must
|
||||
only be called by derived sub-classes, and only from the create() function,
|
||||
only be called by derived sub-classes, and only from the #GstBaseSrcClass::create function,
|
||||
as the stream-lock needs to be held.
|
||||
|
||||
The format for the new segment will be the current format of the source, as
|
||||
|
@ -5425,7 +5619,7 @@ when trying to read more should set this to %FALSE.
|
|||
When @src operates in %GST_FORMAT_TIME, #GstBaseSrc will send an EOS
|
||||
when a buffer outside of the currently configured segment is pushed if
|
||||
@automatic_eos is %TRUE. Since 1.16, if @automatic_eos is %FALSE an
|
||||
EOS will be pushed only when the #GstBaseSrc.create implementation
|
||||
EOS will be pushed only when the #GstBaseSrcClass.create() implementation
|
||||
returns %GST_FLOW_EOS.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -5742,7 +5936,7 @@ buffers.</doc>
|
|||
<parameter name="src" transfer-ownership="none">
|
||||
<type name="BaseSrc" c:type="GstBaseSrc*"/>
|
||||
</parameter>
|
||||
<parameter name="filter" transfer-ownership="none">
|
||||
<parameter name="filter" transfer-ownership="none" nullable="1" allow-none="1">
|
||||
<type name="Gst.Caps" c:type="GstCaps*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
|
@ -5752,10 +5946,12 @@ buffers.</doc>
|
|||
<callback name="negotiate">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="src" transfer-ownership="none">
|
||||
<doc xml:space="preserve">base source instance</doc>
|
||||
<type name="BaseSrc" c:type="GstBaseSrc*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
|
@ -6542,10 +6738,10 @@ It provides for:
|
|||
</parameters>
|
||||
</virtual-method>
|
||||
<method name="get_allocator" c:identifier="gst_base_transform_get_allocator">
|
||||
<doc xml:space="preserve">Lets #GstBaseTransform sub-classes to know the memory @allocator
|
||||
<doc xml:space="preserve">Lets #GstBaseTransform sub-classes know the memory @allocator
|
||||
used by the base class and its @params.
|
||||
|
||||
Unref the @allocator after use it.</doc>
|
||||
Unref the @allocator after use.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
|
@ -6571,7 +6767,7 @@ used</doc>
|
|||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">the instance of the #GstBufferPool used
|
||||
by @trans; free it after use it</doc>
|
||||
by @trans; free it after use</doc>
|
||||
<type name="Gst.BufferPool" c:type="GstBufferPool*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
|
@ -6585,7 +6781,7 @@ by @trans; free it after use it</doc>
|
|||
<doc xml:space="preserve">See if @trans is configured as a in_place transform.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE is the transform is configured in in_place mode.
|
||||
<doc xml:space="preserve">%TRUE if the transform is configured in in_place mode.
|
||||
|
||||
MT safe.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
|
@ -6601,7 +6797,7 @@ MT safe.</doc>
|
|||
<doc xml:space="preserve">See if @trans is configured as a passthrough transform.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE is the transform is configured in passthrough mode.
|
||||
<doc xml:space="preserve">%TRUE if the transform is configured in passthrough mode.
|
||||
|
||||
MT safe.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
|
@ -6629,6 +6825,33 @@ MT safe.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="reconfigure" c:identifier="gst_base_transform_reconfigure" version="1.18">
|
||||
<doc xml:space="preserve">Negotiates src pad caps with downstream elements if the source pad is
|
||||
marked as needing reconfiguring. Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in
|
||||
any case. But marks it again if negotiation fails.
|
||||
|
||||
Do not call this in the #GstBaseTransformClass.transform() or
|
||||
#GstBaseTransformClass.transform_ip() vmethod. Call this in
|
||||
#GstBaseTransformClass.submit_input_buffer(),
|
||||
#GstBaseTransformClass.prepare_output_buffer() or in
|
||||
#GstBaseTransformClass.generate_output() _before_ any output buffer is
|
||||
allocated.
|
||||
|
||||
It will be default be called when handling an ALLOCATION query or at the
|
||||
very beginning of the default #GstBaseTransformClass.submit_input_buffer()
|
||||
implementation.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the negotiation succeeded, else %FALSE.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="trans" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the #GstBaseTransform to set</doc>
|
||||
<type name="BaseTransform" c:type="GstBaseTransform*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="reconfigure_sink" c:identifier="gst_base_transform_reconfigure_sink">
|
||||
<doc xml:space="preserve">Instructs @trans to request renegotiation upstream. This function is
|
||||
typically called after properties on the transform were set that
|
||||
|
@ -6807,14 +7030,14 @@ running_time.</doc>
|
|||
</parameters>
|
||||
</method>
|
||||
<method name="update_src_caps" c:identifier="gst_base_transform_update_src_caps" version="1.6">
|
||||
<doc xml:space="preserve">Updates the srcpad caps and send the caps downstream. This function
|
||||
<doc xml:space="preserve">Updates the srcpad caps and sends the caps downstream. This function
|
||||
can be used by subclasses when they have already negotiated their caps
|
||||
but found a change in them (or computed new information). This way,
|
||||
they can notify downstream about that change without losing any
|
||||
buffer.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the caps could be send downstream %FALSE otherwise</doc>
|
||||
<doc xml:space="preserve">%TRUE if the caps could be sent downstream %FALSE otherwise</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
|
@ -8047,7 +8270,7 @@ Free-function: gst_bit_writer_free</doc>
|
|||
<function name="new_with_data" c:identifier="gst_bit_writer_new_with_data" introspectable="0">
|
||||
<doc xml:space="preserve">Creates a new #GstBitWriter instance with the given memory area. If
|
||||
@initialized is %TRUE it is possible to read @size bits from the
|
||||
#GstBitWriter from the beginnig.
|
||||
#GstBitWriter from the beginning.
|
||||
|
||||
Free-function: gst_bit_writer_free</doc>
|
||||
|
||||
|
@ -12855,7 +13078,7 @@ Free-function: gst_bit_writer_free</doc>
|
|||
<function name="bit_writer_new_with_data" c:identifier="gst_bit_writer_new_with_data" moved-to="BitWriter.new_with_data" introspectable="0">
|
||||
<doc xml:space="preserve">Creates a new #GstBitWriter instance with the given memory area. If
|
||||
@initialized is %TRUE it is possible to read @size bits from the
|
||||
#GstBitWriter from the beginnig.
|
||||
#GstBitWriter from the beginning.
|
||||
|
||||
Free-function: gst_bit_writer_free</doc>
|
||||
|
||||
|
|
|
@ -787,6 +787,27 @@ MT safe.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="pull_until_eos" c:identifier="gst_harness_pull_until_eos" version="1.18">
|
||||
<doc xml:space="preserve">Pulls a #GstBuffer from the #GAsyncQueue on the #GstHarness sinkpad. The pull
|
||||
will block until an EOS event is received, or timeout in 60 seconds.
|
||||
MT safe.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE on success, %FALSE on timeout.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="h" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstHarness</doc>
|
||||
<type name="Harness" c:type="GstHarness*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="buf" direction="out" caller-allocates="0" transfer-ownership="full">
|
||||
<doc xml:space="preserve">A #GstBuffer, or %NULL if EOS or timeout occures
|
||||
first.</doc>
|
||||
<type name="Gst.Buffer" c:type="GstBuffer**"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="pull_upstream_event" c:identifier="gst_harness_pull_upstream_event" version="1.6">
|
||||
<doc xml:space="preserve">Pulls an #GstEvent from the #GAsyncQueue on the #GstHarness srcpad.
|
||||
Timeouts after 60 seconds similar to gst_harness_pull.
|
||||
|
@ -2325,7 +2346,8 @@ MT safe.</doc>
|
|||
<method name="crank" c:identifier="gst_test_clock_crank" version="1.8">
|
||||
<doc xml:space="preserve">A "crank" consists of three steps:
|
||||
1: Wait for a #GstClockID to be registered with the #GstTestClock.
|
||||
2: Advance the #GstTestClock to the time the #GstClockID is waiting for.
|
||||
2: Advance the #GstTestClock to the time the #GstClockID is waiting, unless
|
||||
the clock time is already passed the clock id (Since 1.18).
|
||||
3: Release the #GstClockID wait.
|
||||
A "crank" can be though of as the notion of
|
||||
manually driving the clock forward to its next logical step.</doc>
|
||||
|
@ -2423,6 +2445,25 @@ notification to look for</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="process_id" c:identifier="gst_test_clock_process_id" version="1.18">
|
||||
<doc xml:space="preserve">Processes and releases the pending ID.
|
||||
|
||||
MT safe.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="test_clock" transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GstTestClock for which to process the pending IDs</doc>
|
||||
<type name="TestClock" c:type="GstTestClock*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="pending_id" transfer-ownership="full">
|
||||
<doc xml:space="preserve">#GstClockID</doc>
|
||||
<type name="Gst.ClockID" c:type="GstClockID"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="process_id_list" c:identifier="gst_test_clock_process_id_list" version="1.4">
|
||||
<doc xml:space="preserve">Processes and releases the pending IDs in the list.
|
||||
|
||||
|
|
File diff suppressed because it is too large
Load diff
|
@ -152,6 +152,12 @@ Consult the relevant specifications for more details.</doc>
|
|||
<type name="Descriptor"/>
|
||||
</array>
|
||||
</field>
|
||||
<constructor name="new" c:identifier="gst_mpegts_atsc_mgt_new">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="AtscMGT" c:type="GstMpegtsAtscMGT*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
</record>
|
||||
<record name="AtscMGTTable" c:type="GstMpegtsAtscMGTTable" glib:type-name="GstMpegtsAtscMGTTable" glib:get-type="gst_mpegts_atsc_mgt_table_get_type" c:symbol-prefix="atsc_mgt_table">
|
||||
<doc xml:space="preserve">Source from a @GstMpegtsAtscMGT</doc>
|
||||
|
@ -203,6 +209,88 @@ Consult the relevant specifications for more details.</doc>
|
|||
</array>
|
||||
</field>
|
||||
</record>
|
||||
<record name="AtscRRT" c:type="GstMpegtsAtscRRT" version="1.18" glib:type-name="GstMpegtsAtscRRT" glib:get-type="gst_mpegts_atsc_rrt_get_type" c:symbol-prefix="atsc_rrt">
|
||||
<doc xml:space="preserve">Region Rating Table (A65)</doc>
|
||||
|
||||
<field name="protocol_version" writable="1">
|
||||
<doc xml:space="preserve">The protocol version</doc>
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</field>
|
||||
<field name="names" writable="1">
|
||||
<doc xml:space="preserve">the names</doc>
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="AtscMultString"/>
|
||||
</array>
|
||||
</field>
|
||||
<field name="dimensions_defined" writable="1">
|
||||
<doc xml:space="preserve">the number of dimensions defined for this rating table</doc>
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</field>
|
||||
<field name="dimensions" writable="1">
|
||||
<doc xml:space="preserve">A set of dimensions</doc>
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="AtscRRTDimension"/>
|
||||
</array>
|
||||
</field>
|
||||
<field name="descriptors" writable="1">
|
||||
<doc xml:space="preserve">descriptors</doc>
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</array>
|
||||
</field>
|
||||
<constructor name="new" c:identifier="gst_mpegts_atsc_rrt_new">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="AtscRRT" c:type="GstMpegtsAtscRRT*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
</record>
|
||||
<record name="AtscRRTDimension" c:type="GstMpegtsAtscRRTDimension" glib:type-name="GstMpegtsAtscRRTDimension" glib:get-type="gst_mpegts_atsc_rrt_dimension_get_type" c:symbol-prefix="atsc_rrt_dimension">
|
||||
|
||||
<field name="names" writable="1">
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</array>
|
||||
</field>
|
||||
<field name="graduated_scale" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="values_defined" writable="1">
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</field>
|
||||
<field name="values" writable="1">
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</array>
|
||||
</field>
|
||||
<constructor name="new" c:identifier="gst_mpegts_atsc_rrt_dimension_new">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="AtscRRTDimension" c:type="GstMpegtsAtscRRTDimension*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
</record>
|
||||
<record name="AtscRRTDimensionValue" c:type="GstMpegtsAtscRRTDimensionValue" version="1.18" glib:type-name="GstMpegtsAtscRRTDimensionValue" glib:get-type="gst_mpegts_atsc_rrt_dimension_value_get_type" c:symbol-prefix="atsc_rrt_dimension_value">
|
||||
|
||||
<field name="abbrev_ratings" writable="1">
|
||||
<doc xml:space="preserve">the abbreviated ratings</doc>
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="AtscMultString"/>
|
||||
</array>
|
||||
</field>
|
||||
<field name="ratings" writable="1">
|
||||
<doc xml:space="preserve">the ratings</doc>
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="AtscMultString"/>
|
||||
</array>
|
||||
</field>
|
||||
<constructor name="new" c:identifier="gst_mpegts_atsc_rrt_dimension_value_new">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="AtscRRTDimensionValue" c:type="GstMpegtsAtscRRTDimensionValue*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
</record>
|
||||
<record name="AtscSTT" c:type="GstMpegtsAtscSTT" glib:type-name="GstMpegtsAtscSTT" glib:get-type="gst_mpegts_atsc_stt_get_type" c:symbol-prefix="atsc_stt">
|
||||
<doc xml:space="preserve">System Time Table (A65)</doc>
|
||||
|
||||
|
@ -239,6 +327,12 @@ Consult the relevant specifications for more details.</doc>
|
|||
<doc xml:space="preserve">The UTC date and time</doc>
|
||||
<type name="Gst.DateTime" c:type="GstDateTime*"/>
|
||||
</field>
|
||||
<constructor name="new" c:identifier="gst_mpegts_atsc_stt_new">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="AtscSTT" c:type="GstMpegtsAtscSTT*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
<method name="get_datetime_utc" c:identifier="gst_mpegts_atsc_stt_get_datetime_utc">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
|
@ -284,6 +378,26 @@ Consult the relevant specifications for more details.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_string" c:identifier="gst_mpegts_atsc_string_segment_set_string">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="seg" transfer-ownership="none">
|
||||
<type name="AtscStringSegment" c:type="GstMpegtsAtscStringSegment*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="string" transfer-ownership="none">
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</parameter>
|
||||
<parameter name="compression_type" transfer-ownership="none">
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</parameter>
|
||||
<parameter name="mode" transfer-ownership="none">
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
</record>
|
||||
<record name="AtscVCT" c:type="GstMpegtsAtscVCT" glib:type-name="GstMpegtsAtscVCT" glib:get-type="gst_mpegts_atsc_vct_get_type" c:symbol-prefix="atsc_vct">
|
||||
<doc xml:space="preserve">Represents both:
|
||||
|
@ -739,6 +853,8 @@ Consult the relevant specifications for more details.</doc>
|
|||
</member>
|
||||
<member name="uri_linkage" value="19" c:identifier="GST_MTS_DESC_EXT_DVB_URI_LINKAGE">
|
||||
</member>
|
||||
<member name="ac4" value="21" c:identifier="GST_MTS_DESC_EXT_DVB_AC4">
|
||||
</member>
|
||||
</enumeration>
|
||||
<record name="DVBLinkageDescriptor" c:type="GstMpegtsDVBLinkageDescriptor" glib:type-name="GstMpegtsDVBLinkageDescriptor" glib:get-type="gst_mpegts_dvb_linkage_descriptor_get_type" c:symbol-prefix="dvb_linkage_descriptor">
|
||||
|
||||
|
@ -755,14 +871,14 @@ Consult the relevant specifications for more details.</doc>
|
|||
<type name="guint16" c:type="guint16"/>
|
||||
</field>
|
||||
<field name="linkage_type" writable="1">
|
||||
<doc xml:space="preserve">the type which %linkage_data has</doc>
|
||||
<doc xml:space="preserve">the type which @linkage_data has</doc>
|
||||
<type name="DVBLinkageType" c:type="GstMpegtsDVBLinkageType"/>
|
||||
</field>
|
||||
<field name="linkage_data" readable="0" private="1">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</field>
|
||||
<field name="private_data_length" writable="1">
|
||||
<doc xml:space="preserve">the length for %private_data_bytes</doc>
|
||||
<doc xml:space="preserve">the length for @private_data_bytes</doc>
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</field>
|
||||
<field name="private_data_bytes" writable="1">
|
||||
|
@ -1090,7 +1206,7 @@ As specified in Table 100 of ETSI EN 300 468 v1.13.1</doc>
|
|||
<doc xml:space="preserve">These are the base descriptor types and methods.
|
||||
|
||||
For more details, refer to the ITU H.222.0 or ISO/IEC 13818-1 specifications
|
||||
and other specifications mentionned in the documentation.</doc>
|
||||
and other specifications mentioned in the documentation.</doc>
|
||||
|
||||
<field name="tag" writable="1">
|
||||
<doc xml:space="preserve">the type of descriptor</doc>
|
||||
|
@ -1295,7 +1411,7 @@ are found in http://www.dvbservices.com</doc>
|
|||
</array>
|
||||
</parameter>
|
||||
<parameter name="len" direction="out" caller-allocates="0" transfer-ownership="full">
|
||||
<doc xml:space="preserve">the length of #id_selector_bytes</doc>
|
||||
<doc xml:space="preserve">the length of @id_selector_bytes</doc>
|
||||
<type name="guint8" c:type="guint8*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
|
@ -1336,7 +1452,7 @@ are found in http://www.dvbservices.com</doc>
|
|||
</parameter>
|
||||
<parameter name="list" direction="out" caller-allocates="0" transfer-ownership="full">
|
||||
<doc xml:space="preserve">a list of all frequencies in Hz/kHz
|
||||
depending on %offset</doc>
|
||||
depending on @offset</doc>
|
||||
<array name="GLib.Array" c:type="GArray**">
|
||||
<type name="guint32"/>
|
||||
</array>
|
||||
|
@ -1516,7 +1632,7 @@ registered by http://www.dvbservices.com/</doc>
|
|||
</array>
|
||||
</parameter>
|
||||
<parameter name="length" direction="out" caller-allocates="0" transfer-ownership="full" optional="1" allow-none="1">
|
||||
<doc xml:space="preserve">length of %private_data</doc>
|
||||
<doc xml:space="preserve">length of @private_data</doc>
|
||||
<type name="guint8" c:type="guint8*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
|
@ -1620,7 +1736,7 @@ the list of services</doc>
|
|||
<doc xml:space="preserve">Extracts the component tag from @descriptor.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if the parsing happended correctly, else %FALSE.</doc>
|
||||
<doc xml:space="preserve">%TRUE if the parsing happened correctly, else %FALSE.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
|
@ -2643,6 +2759,135 @@ Corresponds to table 6 of ETSI EN 300 468 (v1.13.0)</doc>
|
|||
<member name="off_air" value="5" c:identifier="GST_MPEGTS_RUNNING_STATUS_OFF_AIR">
|
||||
</member>
|
||||
</enumeration>
|
||||
<record name="SCTESIT" c:type="GstMpegtsSCTESIT" glib:type-name="GstMpegtsSCTESIT" glib:get-type="gst_mpegts_scte_sit_get_type" c:symbol-prefix="scte_sit">
|
||||
|
||||
<field name="encrypted_packet" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="encryption_algorithm" writable="1">
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</field>
|
||||
<field name="pts_adjustment" writable="1">
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</field>
|
||||
<field name="cw_index" writable="1">
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</field>
|
||||
<field name="tier" writable="1">
|
||||
<type name="guint16" c:type="guint16"/>
|
||||
</field>
|
||||
<field name="splice_command_length" writable="1">
|
||||
<type name="guint16" c:type="guint16"/>
|
||||
</field>
|
||||
<field name="splice_command_type" writable="1">
|
||||
<type name="SCTESpliceCommandType" c:type="GstMpegtsSCTESpliceCommandType"/>
|
||||
</field>
|
||||
<field name="splice_time_specified" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="splice_time" writable="1">
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</field>
|
||||
<field name="splices" writable="1">
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</array>
|
||||
</field>
|
||||
<field name="descriptors" writable="1">
|
||||
<array name="GLib.PtrArray" c:type="GPtrArray*">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</array>
|
||||
</field>
|
||||
<constructor name="new" c:identifier="gst_mpegts_scte_sit_new">
|
||||
<doc xml:space="preserve">Allocates and initializes a #GstMpegtsSCTESIT.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
|
||||
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
</record>
|
||||
<enumeration name="SCTESpliceCommandType" c:type="GstMpegtsSCTESpliceCommandType">
|
||||
|
||||
<member name="null" value="0" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_NULL">
|
||||
</member>
|
||||
<member name="schedule" value="4" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_SCHEDULE">
|
||||
</member>
|
||||
<member name="insert" value="5" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_INSERT">
|
||||
</member>
|
||||
<member name="time" value="6" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_TIME">
|
||||
</member>
|
||||
<member name="bandwidth" value="7" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_BANDWIDTH">
|
||||
</member>
|
||||
<member name="private" value="255" c:identifier="GST_MTS_SCTE_SPLICE_COMMAND_PRIVATE">
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="SCTESpliceDescriptor" c:type="GstMpegtsSCTESpliceDescriptor">
|
||||
|
||||
<member name="avail" value="0" c:identifier="GST_MTS_SCTE_DESC_AVAIL">
|
||||
</member>
|
||||
<member name="dtmf" value="1" c:identifier="GST_MTS_SCTE_DESC_DTMF">
|
||||
</member>
|
||||
<member name="segmentation" value="2" c:identifier="GST_MTS_SCTE_DESC_SEGMENTATION">
|
||||
</member>
|
||||
<member name="time" value="3" c:identifier="GST_MTS_SCTE_DESC_TIME">
|
||||
</member>
|
||||
<member name="audio" value="4" c:identifier="GST_MTS_SCTE_DESC_AUDIO">
|
||||
</member>
|
||||
</enumeration>
|
||||
<record name="SCTESpliceEvent" c:type="GstMpegtsSCTESpliceEvent" glib:type-name="GstMpegtsSCTESpliceEvent" glib:get-type="gst_mpegts_scte_splice_event_get_type" c:symbol-prefix="scte_splice_event">
|
||||
|
||||
<field name="insert_event" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="splice_event_id" writable="1">
|
||||
<type name="guint32" c:type="guint32"/>
|
||||
</field>
|
||||
<field name="splice_event_cancel_indicator" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="out_of_network_indicator" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="program_splice_flag" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="duration_flag" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="splice_immediate_flag" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="program_splice_time_specified" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="program_splice_time" writable="1">
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</field>
|
||||
<field name="break_duration_auto_return" writable="1">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="break_duration" writable="1">
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</field>
|
||||
<field name="unique_program_id" writable="1">
|
||||
<type name="guint16" c:type="guint16"/>
|
||||
</field>
|
||||
<field name="avail_num" writable="1">
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</field>
|
||||
<field name="avails_expected" writable="1">
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</field>
|
||||
<constructor name="new" c:identifier="gst_mpegts_scte_splice_event_new">
|
||||
<doc xml:space="preserve">Allocates and initializes a #GstMpegtsSCTESpliceEvent.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESpliceEvent</doc>
|
||||
<type name="SCTESpliceEvent" c:type="GstMpegtsSCTESpliceEvent*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
</record>
|
||||
<record name="SDT" c:type="GstMpegtsSDT" glib:type-name="GstMpegtsSDT" glib:get-type="gst_mpegts_sdt_get_type" c:symbol-prefix="sdt">
|
||||
<doc xml:space="preserve">Service Description Table (EN 300 468)</doc>
|
||||
|
||||
|
@ -2798,6 +3043,9 @@ else in the western part.</doc>
|
|||
<member name="isoch_data" value="131" c:identifier="GST_MPEGTS_STREAM_TYPE_SCTE_ISOCH_DATA">
|
||||
<doc xml:space="preserve">SCTE-19 Isochronous data</doc>
|
||||
</member>
|
||||
<member name="sit" value="134" c:identifier="GST_MPEGTS_STREAM_TYPE_SCTE_SIT">
|
||||
<doc xml:space="preserve">SCTE-35 Splice Information Table</doc>
|
||||
</member>
|
||||
<member name="dst_nrt" value="149" c:identifier="GST_MPEGTS_STREAM_TYPE_SCTE_DST_NRT">
|
||||
<doc xml:space="preserve">SCTE-07 Data Service or
|
||||
Network Resource Table</doc>
|
||||
|
@ -2980,6 +3228,21 @@ happened.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_atsc_rrt" c:identifier="gst_mpegts_section_get_atsc_rrt" version="1.18">
|
||||
<doc xml:space="preserve">Returns the #GstMpegtsAtscRRT contained in the @section.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">The #GstMpegtsAtscRRT contained in the section, or %NULL if an error
|
||||
happened.</doc>
|
||||
<type name="AtscRRT" c:type="const GstMpegtsAtscRRT*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="section" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstMpegtsSection of type %GST_MPEGTS_SECTION_ATSC_RRT</doc>
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_atsc_stt" c:identifier="gst_mpegts_section_get_atsc_stt">
|
||||
<doc xml:space="preserve">Returns the #GstMpegtsAtscSTT contained in the @section.</doc>
|
||||
|
||||
|
@ -3126,6 +3389,21 @@ happened.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_scte_sit" c:identifier="gst_mpegts_section_get_scte_sit">
|
||||
<doc xml:space="preserve">Returns the #GstMpegtsSCTESIT contained in the @section.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">The #GstMpegtsSCTESIT contained in the section, or %NULL if an error
|
||||
happened.</doc>
|
||||
<type name="SCTESIT" c:type="const GstMpegtsSCTESIT*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="section" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstMpegtsSection of type %GST_MPEGTS_SECTION_SCTE_SIT</doc>
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_sdt" c:identifier="gst_mpegts_section_get_sdt">
|
||||
<doc xml:space="preserve">Returns the #GstMpegtsSDT contained in the @section.</doc>
|
||||
|
||||
|
@ -3227,6 +3505,41 @@ The #GstEvent is sent to the @element #GstElement.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<function name="from_atsc_mgt" c:identifier="gst_mpegts_section_from_atsc_mgt" version="1.18">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">the #GstMpegtsSection</doc>
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="mgt" transfer-ownership="full">
|
||||
<doc xml:space="preserve">a #GstMpegtsAtscMGT to create the #GstMpegtsSection from</doc>
|
||||
<type name="AtscMGT" c:type="GstMpegtsAtscMGT*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="from_atsc_rrt" c:identifier="gst_mpegts_section_from_atsc_rrt">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="rrt" transfer-ownership="none">
|
||||
<type name="AtscRRT" c:type="GstMpegtsAtscRRT*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="from_atsc_stt" c:identifier="gst_mpegts_section_from_atsc_stt">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="stt" transfer-ownership="none">
|
||||
<type name="AtscSTT" c:type="GstMpegtsAtscSTT*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="from_nit" c:identifier="gst_mpegts_section_from_nit">
|
||||
<doc xml:space="preserve">Ownership of @nit is taken. The data in @nit is managed by the #GstMpegtsSection</doc>
|
||||
|
||||
|
@ -3279,6 +3592,23 @@ The #GstEvent is sent to the @element #GstElement.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="from_scte_sit" c:identifier="gst_mpegts_section_from_scte_sit">
|
||||
<doc xml:space="preserve">Ownership of @sit is taken. The data in @sit is managed by the #GstMpegtsSection</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">the #GstMpegtsSection</doc>
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="sit" transfer-ownership="full">
|
||||
<doc xml:space="preserve">a #GstMpegtsSCTESIT to create the #GstMpegtsSection from</doc>
|
||||
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
|
||||
</parameter>
|
||||
<parameter name="pid" transfer-ownership="none">
|
||||
<type name="guint16" c:type="guint16"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="from_sdt" c:identifier="gst_mpegts_section_from_sdt">
|
||||
<doc xml:space="preserve">Ownership of @sdt is taken. The data in @sdt is managed by the #GstMpegtsSection</doc>
|
||||
|
||||
|
@ -3544,6 +3874,11 @@ see also #GstMpegtsSectionATSCTableID, #GstMpegtsSectionDVBTableID, and
|
|||
<member name="atsc_stt" value="16" c:identifier="GST_MPEGTS_SECTION_ATSC_STT">
|
||||
<doc xml:space="preserve">ATSC System Time Table (A65)</doc>
|
||||
</member>
|
||||
<member name="atsc_rrt" value="17" c:identifier="GST_MPEGTS_SECTION_ATSC_RRT">
|
||||
</member>
|
||||
<member name="scte_sit" value="18" c:identifier="GST_MPEGTS_SECTION_SCTE_SIT">
|
||||
<doc xml:space="preserve">SCTE Splice Information Table (SCTE-35)</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="StreamType" c:type="GstMpegtsStreamType">
|
||||
<doc xml:space="preserve">Type of mpeg-ts stream type.
|
||||
|
@ -3803,7 +4138,7 @@ profiles defined in Annex A for service-compatible stereoscopic 3D services</doc
|
|||
<type name="gboolean" c:type="gboolean"/>
|
||||
</field>
|
||||
<field name="constellation" writable="1">
|
||||
<doc xml:space="preserve">the constallation</doc>
|
||||
<doc xml:space="preserve">the constellation</doc>
|
||||
<type name="ModulationType" c:type="GstMpegtsModulationType"/>
|
||||
</field>
|
||||
<field name="hierarchy" writable="1">
|
||||
|
@ -4073,7 +4408,7 @@ Note: To look for descriptors that can be present more than once in an
|
|||
array of descriptors, iterate the #GArray manually.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">the first descriptor matchin @tag, else %NULL.</doc>
|
||||
<doc xml:space="preserve">the first descriptor matching @tag, else %NULL.</doc>
|
||||
<type name="Descriptor" c:type="const GstMpegtsDescriptor*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
|
@ -4166,6 +4501,115 @@ Release with #g_array_unref when done with it.</doc>
|
|||
</array>
|
||||
</return-value>
|
||||
</function>
|
||||
<function name="scte_cancel_new" c:identifier="gst_mpegts_scte_cancel_new">
|
||||
<doc xml:space="preserve">Allocates and initializes a new INSERT command #GstMpegtsSCTESIT
|
||||
setup to cancel the specified @event_id.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
|
||||
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="event_id" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The event ID to cancel.</doc>
|
||||
<type name="guint32" c:type="guint32"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="scte_null_new" c:identifier="gst_mpegts_scte_null_new">
|
||||
<doc xml:space="preserve">Allocates and initializes a NULL command #GstMpegtsSCTESIT.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
|
||||
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
|
||||
</return-value>
|
||||
</function>
|
||||
<function name="scte_splice_in_new" c:identifier="gst_mpegts_scte_splice_in_new">
|
||||
<doc xml:space="preserve">Allocates and initializes a new "Splice In" INSERT command
|
||||
#GstMpegtsSCTESIT for the given @event_id and @splice_time.
|
||||
|
||||
If the @splice_time is #G_MAXUINT64 then the event will be
|
||||
immediate as opposed to for the target @splice_time.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
|
||||
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="event_id" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The event ID.</doc>
|
||||
<type name="guint32" c:type="guint32"/>
|
||||
</parameter>
|
||||
<parameter name="splice_time" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The PCR time for the splice event</doc>
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="scte_splice_out_new" c:identifier="gst_mpegts_scte_splice_out_new">
|
||||
<doc xml:space="preserve">Allocates and initializes a new "Splice Out" INSERT command
|
||||
#GstMpegtsSCTESIT for the given @event_id, @splice_time and
|
||||
duration.
|
||||
|
||||
If the @splice_time is #G_MAXUINT64 then the event will be
|
||||
immediate as opposed to for the target @splice_time.
|
||||
|
||||
If the @duration is 0 it won't be specified in the event.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">A newly allocated #GstMpegtsSCTESIT</doc>
|
||||
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="event_id" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The event ID.</doc>
|
||||
<type name="guint32" c:type="guint32"/>
|
||||
</parameter>
|
||||
<parameter name="splice_time" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The PCR time for the splice event</doc>
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</parameter>
|
||||
<parameter name="duration" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The optional duration.</doc>
|
||||
<type name="guint64" c:type="guint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="section_from_atsc_mgt" c:identifier="gst_mpegts_section_from_atsc_mgt" moved-to="Section.from_atsc_mgt" version="1.18">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">the #GstMpegtsSection</doc>
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="mgt" transfer-ownership="full">
|
||||
<doc xml:space="preserve">a #GstMpegtsAtscMGT to create the #GstMpegtsSection from</doc>
|
||||
<type name="AtscMGT" c:type="GstMpegtsAtscMGT*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="section_from_atsc_rrt" c:identifier="gst_mpegts_section_from_atsc_rrt" moved-to="Section.from_atsc_rrt">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="rrt" transfer-ownership="none">
|
||||
<type name="AtscRRT" c:type="GstMpegtsAtscRRT*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="section_from_atsc_stt" c:identifier="gst_mpegts_section_from_atsc_stt" moved-to="Section.from_atsc_stt">
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="stt" transfer-ownership="none">
|
||||
<type name="AtscSTT" c:type="GstMpegtsAtscSTT*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="section_from_nit" c:identifier="gst_mpegts_section_from_nit" moved-to="Section.from_nit">
|
||||
<doc xml:space="preserve">Ownership of @nit is taken. The data in @nit is managed by the #GstMpegtsSection</doc>
|
||||
|
||||
|
@ -4218,6 +4662,23 @@ Release with #g_array_unref when done with it.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="section_from_scte_sit" c:identifier="gst_mpegts_section_from_scte_sit" moved-to="Section.from_scte_sit">
|
||||
<doc xml:space="preserve">Ownership of @sit is taken. The data in @sit is managed by the #GstMpegtsSection</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">the #GstMpegtsSection</doc>
|
||||
<type name="Section" c:type="GstMpegtsSection*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="sit" transfer-ownership="full">
|
||||
<doc xml:space="preserve">a #GstMpegtsSCTESIT to create the #GstMpegtsSection from</doc>
|
||||
<type name="SCTESIT" c:type="GstMpegtsSCTESIT*"/>
|
||||
</parameter>
|
||||
<parameter name="pid" transfer-ownership="none">
|
||||
<type name="guint16" c:type="guint16"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="section_from_sdt" c:identifier="gst_mpegts_section_from_sdt" moved-to="Section.from_sdt">
|
||||
<doc xml:space="preserve">Ownership of @sdt is taken. The data in @sdt is managed by the #GstMpegtsSection</doc>
|
||||
|
||||
|
|
|
@ -159,7 +159,7 @@ send @GST_MESSAGE_ELEMENT messages with an attached #GstStructure containing
|
|||
statistics about clock accuracy and network traffic.</doc>
|
||||
|
||||
<constructor name="new" c:identifier="gst_net_client_clock_new">
|
||||
<doc xml:space="preserve">Create a new #GstNetClientInternalClock that will report the time
|
||||
<doc xml:space="preserve">Create a new #GstNetClientClock that will report the time
|
||||
provided by the #GstNetTimeProvider on @remote_address and
|
||||
@remote_port.</doc>
|
||||
|
||||
|
@ -799,6 +799,24 @@ calls, but otherwise returns NULL on error.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="net_utils_set_socket_tos" c:identifier="gst_net_utils_set_socket_tos" version="1.18">
|
||||
<doc xml:space="preserve">Configures IP_TOS value of socket, i.e. sets QoS DSCP.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">TRUE if successful, FALSE in case an error occurred.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="socket" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Socket to configure</doc>
|
||||
<type name="Gio.Socket" c:type="GSocket*"/>
|
||||
</parameter>
|
||||
<parameter name="qos_dscp" transfer-ownership="none">
|
||||
<doc xml:space="preserve">QoS DSCP value</doc>
|
||||
<type name="gint" c:type="gint"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="ptp_deinit" c:identifier="gst_ptp_deinit" version="1.6">
|
||||
<doc xml:space="preserve">Deinitialize the GStreamer PTP subsystem and stop the PTP clock. If there
|
||||
are any remaining GstPtpClock instances, they won't be further synchronized
|
||||
|
|
|
@ -65,7 +65,7 @@ audio-rate to video-rate and handles renegotiation (downstream video size
|
|||
changes).
|
||||
|
||||
It also provides several background shading effects. These effects are
|
||||
applied to a previous picture before the render() implementation can draw a
|
||||
applied to a previous picture before the `render()` implementation can draw a
|
||||
new frame.</doc>
|
||||
|
||||
<virtual-method name="decide_allocation">
|
||||
|
@ -849,7 +849,7 @@ gst_discoverer_stream_info_list_free().</doc>
|
|||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">An array of strings
|
||||
containing informations about how to install the various missing elements
|
||||
containing information about how to install the various missing elements
|
||||
for @info to be usable. If you wish to use the strings after the life-time
|
||||
of @info, you will need to copy them.</doc>
|
||||
<array c:type="const gchar**">
|
||||
|
@ -1807,6 +1807,21 @@ Can be %NULL. Unref after usage.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_single_segment" c:identifier="gst_encoding_profile_get_single_segment" version="1.18">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#TRUE if the stream represented by @profile should use a single
|
||||
segment before the encoder, #FALSE otherwise. This means that buffers will be retimestamped
|
||||
and segments will be eat so as to appear as one segment.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="profile" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstEncodingProfile</doc>
|
||||
<type name="EncodingProfile" c:type="GstEncodingProfile*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_type_nick" c:identifier="gst_encoding_profile_get_type_nick">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -1898,7 +1913,7 @@ during the encoding</doc>
|
|||
<type name="EncodingProfile" c:type="GstEncodingProfile*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="enabled" transfer-ownership="none">
|
||||
<doc xml:space="preserve">%FALSE to disable #profile, %TRUE to enable it</doc>
|
||||
<doc xml:space="preserve">%FALSE to disable @profile, %TRUE to enable it</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
|
@ -2011,6 +2026,25 @@ for more about restrictions. Does not apply to #GstEncodingContainerProfile.</do
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_single_segment" c:identifier="gst_encoding_profile_set_single_segment" version="1.18">
|
||||
<doc xml:space="preserve">If using a single segment, buffers will be retimestamped
|
||||
and segments will be eat so as to appear as one segment.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="profile" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstEncodingProfile</doc>
|
||||
<type name="EncodingProfile" c:type="GstEncodingProfile*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="single_segment" transfer-ownership="none">
|
||||
<doc xml:space="preserve">#TRUE if the stream represented by @profile should use a single
|
||||
segment before the encoder #FALSE otherwise.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<property name="restriction-caps" writable="1" transfer-ownership="none">
|
||||
<type name="Gst.Caps"/>
|
||||
</property>
|
||||
|
@ -2030,9 +2064,9 @@ The name and category can only consist of lowercase ASCII letters for the
|
|||
first character, followed by either lowercase ASCII letters, digits or
|
||||
hyphens ('-').
|
||||
|
||||
The @category <emphasis>should</emphasis> be one of the existing
|
||||
The @category *should* be one of the existing
|
||||
well-defined categories, like #GST_ENCODING_CATEGORY_DEVICE, but it
|
||||
<emphasis>can</emphasis> be a application or user specific category if
|
||||
*can* be a application or user specific category if
|
||||
needed.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
|
@ -2641,7 +2675,7 @@ program and to what extent the requested plugins could be installed.</doc>
|
|||
some (but not all) of the requested plugins could be installed.</doc>
|
||||
</member>
|
||||
<member name="error" value="2" c:identifier="GST_INSTALL_PLUGINS_ERROR" glib:nick="error">
|
||||
<doc xml:space="preserve">an error occured during the installation. If
|
||||
<doc xml:space="preserve">an error occurred during the installation. If
|
||||
this happens, the user has already seen an error message and another
|
||||
one should not be displayed</doc>
|
||||
</member>
|
||||
|
@ -2666,7 +2700,7 @@ program and to what extent the requested plugins could be installed.</doc>
|
|||
</member>
|
||||
<member name="internal_failure" value="201" c:identifier="GST_INSTALL_PLUGINS_INTERNAL_FAILURE" glib:nick="internal-failure">
|
||||
<doc xml:space="preserve">some internal failure has
|
||||
occured when trying to start the installer</doc>
|
||||
occurred when trying to start the installer</doc>
|
||||
</member>
|
||||
<member name="helper_missing" value="202" c:identifier="GST_INSTALL_PLUGINS_HELPER_MISSING" glib:nick="helper-missing">
|
||||
<doc xml:space="preserve">the helper script to call the
|
||||
|
@ -2706,17 +2740,17 @@ in debugging.</doc>
|
|||
|
||||
<type name="gint" c:type="gint"/>
|
||||
</constant>
|
||||
<constant name="PLUGINS_BASE_VERSION_MICRO" value="2" c:type="GST_PLUGINS_BASE_VERSION_MICRO">
|
||||
<constant name="PLUGINS_BASE_VERSION_MICRO" value="0" c:type="GST_PLUGINS_BASE_VERSION_MICRO">
|
||||
<doc xml:space="preserve">The micro version of GStreamer's gst-plugins-base libraries at compile time.</doc>
|
||||
|
||||
<type name="gint" c:type="gint"/>
|
||||
</constant>
|
||||
<constant name="PLUGINS_BASE_VERSION_MINOR" value="16" c:type="GST_PLUGINS_BASE_VERSION_MINOR">
|
||||
<constant name="PLUGINS_BASE_VERSION_MINOR" value="17" c:type="GST_PLUGINS_BASE_VERSION_MINOR">
|
||||
<doc xml:space="preserve">The minor version of GStreamer's gst-plugins-base libraries at compile time.</doc>
|
||||
|
||||
<type name="gint" c:type="gint"/>
|
||||
</constant>
|
||||
<constant name="PLUGINS_BASE_VERSION_NANO" value="0" c:type="GST_PLUGINS_BASE_VERSION_NANO">
|
||||
<constant name="PLUGINS_BASE_VERSION_NANO" value="1" c:type="GST_PLUGINS_BASE_VERSION_NANO">
|
||||
<doc xml:space="preserve">The nano version of GStreamer's gst-plugins-base libraries at compile time.
|
||||
Actual releases have 0, GIT versions have 1, prerelease versions have 2-...</doc>
|
||||
|
||||
|
@ -3063,7 +3097,7 @@ with bit 0 being the most significant bit of the first byte.
|
|||
* Bit 41 - general_interlaced_source_flag
|
||||
* Bit 42 - general_non_packed_constraint_flag
|
||||
* Bit 43 - general_frame_only_constraint_flag
|
||||
* Bit 44:87 - general_reserved_zero_44bits
|
||||
* Bit 44:87 - See below
|
||||
* Bit 88:95 - general_level_idc</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
|
|
@ -1073,7 +1073,7 @@ value.</doc>
|
|||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE or %FALSE
|
||||
|
||||
Sets the subtitle strack @stream_index.</doc>
|
||||
Sets the subtitle stack @stream_index.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
|
@ -1567,7 +1567,7 @@ gain.</doc>
|
|||
<implements name="PlayerSignalDispatcher"/>
|
||||
<function name="new" c:identifier="gst_player_g_main_context_signal_dispatcher_new">
|
||||
<doc xml:space="preserve">Creates a new GstPlayerSignalDispatcher that uses @application_context,
|
||||
or the thread default one if %NULL is used. See gst_player_new_full().</doc>
|
||||
or the thread default one if %NULL is used. See gst_player_new().</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">the new GstPlayerSignalDispatcher</doc>
|
||||
|
@ -1634,7 +1634,7 @@ matching #GstPlayerAudioInfo.</doc>
|
|||
</method>
|
||||
<method name="get_image_sample" c:identifier="gst_player_media_info_get_image_sample">
|
||||
<doc xml:space="preserve">Function to get the image (or preview-image) stored in taglist.
|
||||
Application can use gst_sample_*_() API's to get caps, buffer etc.</doc>
|
||||
Application can use `gst_sample_*_()` API's to get caps, buffer etc.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">GstSample or NULL.</doc>
|
||||
|
@ -1927,7 +1927,7 @@ stream.</doc>
|
|||
</function>
|
||||
</enumeration>
|
||||
<class name="PlayerStreamInfo" c:symbol-prefix="player_stream_info" c:type="GstPlayerStreamInfo" parent="GObject.Object" abstract="1" glib:type-name="GstPlayerStreamInfo" glib:get-type="gst_player_stream_info_get_type" glib:type-struct="PlayerStreamInfoClass">
|
||||
<doc xml:space="preserve">Base structure for information concering a media stream. Depending on
|
||||
<doc xml:space="preserve">Base structure for information concerning a media stream. Depending on
|
||||
the stream type, one can find more media-specific information in
|
||||
#GstPlayerVideoInfo, #GstPlayerAudioInfo, #GstPlayerSubtitleInfo.</doc>
|
||||
|
||||
|
|
|
@ -321,6 +321,8 @@ gst_rtcp_buffer_validate_reduced().</doc>
|
|||
<member name="rtpfb_type_rtcp_sr_req" value="5" c:identifier="GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ" glib:nick="rtpfb-type-rtcp-sr-req">
|
||||
<doc xml:space="preserve">Request an SR packet for early
|
||||
synchronization</doc>
|
||||
</member>
|
||||
<member name="rtpfb_type_twcc" value="15" c:identifier="GST_RTCP_RTPFB_TYPE_TWCC" glib:nick="rtpfb-type-twcc">
|
||||
</member>
|
||||
<member name="psfb_type_pli" value="1" c:identifier="GST_RTCP_PSFB_TYPE_PLI" glib:nick="psfb-type-pli">
|
||||
<doc xml:space="preserve">Picture Loss Indication</doc>
|
||||
|
@ -2096,7 +2098,7 @@ RTP packets always contain full frames.
|
|||
To use this base class, your child element needs to call either
|
||||
gst_rtp_base_audio_payload_set_frame_based() or
|
||||
gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the
|
||||
element's _init() function. Then, the child element must call either
|
||||
element's `_init()` function. Then, the child element must call either
|
||||
gst_rtp_base_audio_payload_set_frame_options(),
|
||||
gst_rtp_base_audio_payload_set_sample_options() or
|
||||
gst_rtp_base_audio_payload_set_samplebits_options. Since
|
||||
|
@ -2393,7 +2395,7 @@ audio codec</doc>
|
|||
<doc xml:space="preserve">Push @out_buf to the peer of @filter. This function takes ownership of
|
||||
@out_buf.
|
||||
|
||||
This function will by default apply the last incomming timestamp on
|
||||
This function will by default apply the last incoming timestamp on
|
||||
the outgoing buffer when it didn't have a timestamp already.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -2447,6 +2449,13 @@ the outgoing buffer when it didn't have a timestamp already.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<property name="max-reorder" version="1.18" writable="1" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Max seqnum reorder before the sender is assumed to have restarted.
|
||||
|
||||
When max-reorder is set to 0 all reordered/duplicate packets are
|
||||
considered coming from a restarted sender.</doc>
|
||||
<type name="gint" c:type="gint"/>
|
||||
</property>
|
||||
<property name="source-info" version="1.16" writable="1" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Add RTP source information found in RTP header as meta to output buffer.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
|
@ -2931,6 +2940,17 @@ timestamps for audio streams.</doc>
|
|||
<doc xml:space="preserve">Force buffers to be multiples of this duration in ns (0 disables)</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</property>
|
||||
<property name="scale-rtptime" version="1.18" writable="1" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Make the RTP packets' timestamps be scaled with the segment's rate
|
||||
(corresponding to RTSP speed parameter). Disabling this property means
|
||||
the timestamps will not be affected by the set delivery speed (RTSP speed).
|
||||
|
||||
Example: A server wants to allow streaming a recorded video in double
|
||||
speed but still have the timestamps correspond to the position in the
|
||||
video. This is achieved by the client setting RTSP Speed to 2 while the
|
||||
server has this property disabled.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</property>
|
||||
<property name="seqnum" transfer-ownership="none">
|
||||
<type name="guint" c:type="guint"/>
|
||||
</property>
|
||||
|
@ -2968,6 +2988,16 @@ the last processed buffer and current state of the stream being payloaded:
|
|||
<property name="timestamp-offset" writable="1" transfer-ownership="none">
|
||||
<type name="guint" c:type="guint"/>
|
||||
</property>
|
||||
<property name="twcc-ext-id" version="1.18" writable="1" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The RTP header-extension ID used for tagging buffers with Transport-Wide
|
||||
Congestion Control sequence-numbers.
|
||||
|
||||
To use this across multiple bundled streams (transport wide), the
|
||||
GstRTPFunnel can mux TWCC sequence-numbers together.
|
||||
|
||||
This is experimental, as it is still a draft and not yet a standard.</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</property>
|
||||
<field name="element">
|
||||
<type name="Gst.Element" c:type="GstElement"/>
|
||||
</field>
|
||||
|
@ -3197,7 +3227,7 @@ the last processed buffer and current state of the stream being payloaded:
|
|||
RTP header. If there is already a RFC 5285 header extension with a one byte
|
||||
header, the new extension will be appended.
|
||||
It will not work if there is already a header extension that does not follow
|
||||
the mecanism described in RFC 5285 or if there is a header extension with
|
||||
the mechanism described in RFC 5285 or if there is a header extension with
|
||||
a two bytes header as described in RFC 5285. In that case, use
|
||||
gst_rtp_buffer_add_extension_twobytes_header()</doc>
|
||||
|
||||
|
@ -3231,7 +3261,7 @@ gst_rtp_buffer_add_extension_twobytes_header()</doc>
|
|||
RTP header. If there is already a RFC 5285 header extension with a two bytes
|
||||
header, the new extension will be appended.
|
||||
It will not work if there is already a header extension that does not follow
|
||||
the mecanism described in RFC 5285 or if there is a header extension with
|
||||
the mechanism described in RFC 5285 or if there is a header extension with
|
||||
a one byte header as described in RFC 5285. In that case, use
|
||||
gst_rtp_buffer_add_extension_onebyte_header()</doc>
|
||||
|
||||
|
@ -4039,6 +4069,46 @@ into account:
|
|||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="get_extension_onebyte_header_from_bytes" c:identifier="gst_rtp_buffer_get_extension_onebyte_header_from_bytes" version="1.18">
|
||||
<doc xml:space="preserve">Similar to gst_rtp_buffer_get_extension_onebyte_header, but working
|
||||
on the #GBytes you get from gst_rtp_buffer_get_extension_bytes.
|
||||
Parses RFC 5285 style header extensions with a one byte header. It will
|
||||
return the nth extension with the requested id.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">TRUE if @bytes had the requested header extension</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="bytes" transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GBytes</doc>
|
||||
<type name="GLib.Bytes" c:type="GBytes*"/>
|
||||
</parameter>
|
||||
<parameter name="bit_pattern" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The bit-pattern. Anything but 0xBEDE is rejected.</doc>
|
||||
<type name="guint16" c:type="guint16"/>
|
||||
</parameter>
|
||||
<parameter name="id" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The ID of the header extension to be read (between 1 and 14).</doc>
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</parameter>
|
||||
<parameter name="nth" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Read the nth extension packet with the requested ID</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</parameter>
|
||||
<parameter name="data" direction="out" caller-allocates="0" transfer-ownership="none">
|
||||
<doc xml:space="preserve">
|
||||
location for data</doc>
|
||||
<array length="5" zero-terminated="0" c:type="gpointer*">
|
||||
<type name="guint8"/>
|
||||
</array>
|
||||
</parameter>
|
||||
<parameter name="size" direction="out" caller-allocates="0" transfer-ownership="full">
|
||||
<doc xml:space="preserve">the size of the data in bytes</doc>
|
||||
<type name="guint" c:type="guint*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="map" c:identifier="gst_rtp_buffer_map">
|
||||
<doc xml:space="preserve">Map the contents of @buffer into @rtp.</doc>
|
||||
|
||||
|
@ -4320,7 +4390,7 @@ channels. NULL = not applicable.</doc>
|
|||
mostly used to get the default clock-rate and bandwidth for dynamic payload
|
||||
types specified with @media and @encoding name.
|
||||
|
||||
The search for @encoding_name will be performed in a case insensitve way.</doc>
|
||||
The search for @encoding_name will be performed in a case insensitive way.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTPPayloadInfo or NULL when no info could be found.</doc>
|
||||
|
@ -5180,6 +5250,46 @@ into account:
|
|||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="rtp_buffer_get_extension_onebyte_header_from_bytes" c:identifier="gst_rtp_buffer_get_extension_onebyte_header_from_bytes" moved-to="RTPBuffer.get_extension_onebyte_header_from_bytes" version="1.18">
|
||||
<doc xml:space="preserve">Similar to gst_rtp_buffer_get_extension_onebyte_header, but working
|
||||
on the #GBytes you get from gst_rtp_buffer_get_extension_bytes.
|
||||
Parses RFC 5285 style header extensions with a one byte header. It will
|
||||
return the nth extension with the requested id.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">TRUE if @bytes had the requested header extension</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="bytes" transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GBytes</doc>
|
||||
<type name="GLib.Bytes" c:type="GBytes*"/>
|
||||
</parameter>
|
||||
<parameter name="bit_pattern" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The bit-pattern. Anything but 0xBEDE is rejected.</doc>
|
||||
<type name="guint16" c:type="guint16"/>
|
||||
</parameter>
|
||||
<parameter name="id" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The ID of the header extension to be read (between 1 and 14).</doc>
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</parameter>
|
||||
<parameter name="nth" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Read the nth extension packet with the requested ID</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</parameter>
|
||||
<parameter name="data" direction="out" caller-allocates="0" transfer-ownership="none">
|
||||
<doc xml:space="preserve">
|
||||
location for data</doc>
|
||||
<array length="5" zero-terminated="0" c:type="gpointer*">
|
||||
<type name="guint8"/>
|
||||
</array>
|
||||
</parameter>
|
||||
<parameter name="size" direction="out" caller-allocates="0" transfer-ownership="full">
|
||||
<doc xml:space="preserve">the size of the data in bytes</doc>
|
||||
<type name="guint" c:type="guint*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</function>
|
||||
<function name="rtp_buffer_map" c:identifier="gst_rtp_buffer_map" moved-to="RTPBuffer.map">
|
||||
<doc xml:space="preserve">Map the contents of @buffer into @rtp.</doc>
|
||||
|
||||
|
@ -5399,7 +5509,7 @@ extension. @data must hold at least #GST_RTP_HDREXT_NTP_64_SIZE bytes.</doc>
|
|||
mostly used to get the default clock-rate and bandwidth for dynamic payload
|
||||
types specified with @media and @encoding name.
|
||||
|
||||
The search for @encoding_name will be performed in a case insensitve way.</doc>
|
||||
The search for @encoding_name will be performed in a case insensitive way.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTPPayloadInfo or NULL when no info could be found.</doc>
|
||||
|
|
|
@ -111,7 +111,7 @@ state as when it was first created.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="connect" c:identifier="gst_rtsp_connection_connect">
|
||||
<method name="connect" c:identifier="gst_rtsp_connection_connect" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Attempt to connect to the url of @conn made with
|
||||
gst_rtsp_connection_create(). If @timeout is %NULL this function can block
|
||||
forever. If @timeout contains a valid timeout, this function will return
|
||||
|
@ -129,12 +129,35 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GTimeVal timeout</doc>
|
||||
<doc xml:space="preserve">a GTimeVal timeout</doc>
|
||||
<type name="GLib.TimeVal" c:type="GTimeVal*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="connect_with_response" c:identifier="gst_rtsp_connection_connect_with_response" version="1.8">
|
||||
<method name="connect_usec" c:identifier="gst_rtsp_connection_connect_usec" version="1.18">
|
||||
<doc xml:space="preserve">Attempt to connect to the url of @conn made with
|
||||
gst_rtsp_connection_create(). If @timeout is 0 this function can block
|
||||
forever. If @timeout contains a valid timeout, this function will return
|
||||
#GST_RTSP_ETIMEOUT after the timeout expired.
|
||||
|
||||
This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GST_RTSP_OK when a connection could be made.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout in microseconds</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="connect_with_response" c:identifier="gst_rtsp_connection_connect_with_response" version="1.8" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Attempt to connect to the url of @conn made with
|
||||
gst_rtsp_connection_create(). If @timeout is %NULL this function can block
|
||||
forever. If @timeout contains a valid timeout, this function will return
|
||||
|
@ -153,7 +176,7 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GTimeVal timeout</doc>
|
||||
<doc xml:space="preserve">a GTimeVal timeout</doc>
|
||||
<type name="GLib.TimeVal" c:type="GTimeVal*"/>
|
||||
</parameter>
|
||||
<parameter name="response" transfer-ownership="none">
|
||||
|
@ -162,6 +185,34 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="connect_with_response_usec" c:identifier="gst_rtsp_connection_connect_with_response_usec" version="1.18">
|
||||
<doc xml:space="preserve">Attempt to connect to the url of @conn made with
|
||||
gst_rtsp_connection_create(). If @timeout is 0 this function can block
|
||||
forever. If @timeout contains a valid timeout, this function will return
|
||||
#GST_RTSP_ETIMEOUT after the timeout expired. If @conn is set to tunneled,
|
||||
@response will contain a response to the tunneling request messages.
|
||||
|
||||
This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GST_RTSP_OK when a connection could be made.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout in microseconds</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
<parameter name="response" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPMessage</doc>
|
||||
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="do_tunnel" c:identifier="gst_rtsp_connection_do_tunnel">
|
||||
<doc xml:space="preserve">If @conn received the first tunnel connection and @conn2 received
|
||||
the second tunnel connection, link the two connections together so that
|
||||
|
@ -395,7 +446,7 @@ error. The file descriptor remains valid until the connection is closed.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="next_timeout" c:identifier="gst_rtsp_connection_next_timeout">
|
||||
<method name="next_timeout" c:identifier="gst_rtsp_connection_next_timeout" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Calculate the next timeout for @conn, storing the result in @timeout.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -413,7 +464,21 @@ error. The file descriptor remains valid until the connection is closed.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="poll" c:identifier="gst_rtsp_connection_poll">
|
||||
<method name="next_timeout_usec" c:identifier="gst_rtsp_connection_next_timeout_usec" version="1.18">
|
||||
<doc xml:space="preserve">Calculate the next timeout for @conn</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#the next timeout in microseconds</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="poll" c:identifier="gst_rtsp_connection_poll" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Wait up to the specified @timeout for the connection to become available for
|
||||
at least one of the operations specified in @events. When the function returns
|
||||
with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
|
||||
|
@ -446,7 +511,40 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="read" c:identifier="gst_rtsp_connection_read">
|
||||
<method name="poll_usec" c:identifier="gst_rtsp_connection_poll_usec" version="1.18">
|
||||
<doc xml:space="preserve">Wait up to the specified @timeout for the connection to become available for
|
||||
at least one of the operations specified in @events. When the function returns
|
||||
with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
|
||||
@conn.
|
||||
|
||||
@timeout can be 0, in which case this function might block forever.
|
||||
|
||||
This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GST_RTSP_OK on success.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="events" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a bitmask of #GstRTSPEvent flags to check</doc>
|
||||
<type name="RTSPEvent" c:type="GstRTSPEvent"/>
|
||||
</parameter>
|
||||
<parameter name="revents" transfer-ownership="none">
|
||||
<doc xml:space="preserve">location for result flags</doc>
|
||||
<type name="RTSPEvent" c:type="GstRTSPEvent*"/>
|
||||
</parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout in microseconds</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="read" c:identifier="gst_rtsp_connection_read" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Attempt to read @size bytes into @data from the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be %NULL, in which case this function
|
||||
might block forever.
|
||||
|
@ -476,7 +574,37 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="receive" c:identifier="gst_rtsp_connection_receive">
|
||||
<method name="read_usec" c:identifier="gst_rtsp_connection_read_usec" version="1.18">
|
||||
<doc xml:space="preserve">Attempt to read @size bytes into @data from the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be 0, in which case this function
|
||||
might block forever.
|
||||
|
||||
This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GST_RTSP_OK on success.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="data" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the data to read</doc>
|
||||
<type name="guint8" c:type="guint8*"/>
|
||||
</parameter>
|
||||
<parameter name="size" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the size of @data</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout value in microseconds</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="receive" c:identifier="gst_rtsp_connection_receive" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Attempt to read into @message from the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be %NULL, in which case this function
|
||||
might block forever.
|
||||
|
@ -502,6 +630,32 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="receive_usec" c:identifier="gst_rtsp_connection_receive_usec" version="1.18">
|
||||
<doc xml:space="preserve">Attempt to read into @message from the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be 0, in which case this function
|
||||
might block forever.
|
||||
|
||||
This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GST_RTSP_OK on success.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="message" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the message to read</doc>
|
||||
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
|
||||
</parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout value or 0</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="reset_timeout" c:identifier="gst_rtsp_connection_reset_timeout">
|
||||
<doc xml:space="preserve">Reset the timeout of @conn.</doc>
|
||||
|
||||
|
@ -516,7 +670,7 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="send" c:identifier="gst_rtsp_connection_send">
|
||||
<method name="send" c:identifier="gst_rtsp_connection_send" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Attempt to send @message to the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be %NULL, in which case this function
|
||||
might block forever.
|
||||
|
@ -542,7 +696,7 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="send_messages" c:identifier="gst_rtsp_connection_send_messages" version="1.16">
|
||||
<method name="send_messages" c:identifier="gst_rtsp_connection_send_messages" version="1.16" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Attempt to send @messages to the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be %NULL, in which case this function
|
||||
might block forever.
|
||||
|
@ -574,6 +728,64 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="send_messages_usec" c:identifier="gst_rtsp_connection_send_messages_usec" version="1.18">
|
||||
<doc xml:space="preserve">Attempt to send @messages to the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be 0, in which case this function
|
||||
might block forever.
|
||||
|
||||
This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GST_RTSP_OK on Since.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="messages" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the messages to send</doc>
|
||||
<array length="1" zero-terminated="0" c:type="GstRTSPMessage*">
|
||||
<type name="RTSPMessage" c:type="GstRTSPMessage"/>
|
||||
</array>
|
||||
</parameter>
|
||||
<parameter name="n_messages" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the number of messages to send</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout value in microseconds</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="send_usec" c:identifier="gst_rtsp_connection_send_usec" version="1.18">
|
||||
<doc xml:space="preserve">Attempt to send @message to the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be 0, in which case this function
|
||||
might block forever.
|
||||
|
||||
This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GST_RTSP_OK on success.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="message" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the message to send</doc>
|
||||
<type name="RTSPMessage" c:type="GstRTSPMessage*"/>
|
||||
</parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout value in microseconds</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_accept_certificate_func" c:identifier="gst_rtsp_connection_set_accept_certificate_func" version="1.14">
|
||||
<doc xml:space="preserve">Sets a custom accept-certificate function for checking certificates for
|
||||
validity. This will directly map to #GTlsConnection 's "accept-certificate"
|
||||
|
@ -632,7 +844,7 @@ user and password respectively.</doc>
|
|||
</parameters>
|
||||
</method>
|
||||
<method name="set_auth_param" c:identifier="gst_rtsp_connection_set_auth_param">
|
||||
<doc xml:space="preserve">Setup @conn with authentication directives. This is not necesary for
|
||||
<doc xml:space="preserve">Setup @conn with authentication directives. This is not necessary for
|
||||
methods #GST_RTSP_AUTH_NONE and #GST_RTSP_AUTH_BASIC. For
|
||||
#GST_RTSP_AUTH_DIGEST, directives should be taken from the digest challenge
|
||||
in the WWW-Authenticate response header and can include realm, domain,
|
||||
|
@ -656,6 +868,25 @@ nonce, opaque, stale, algorithm, qop as per RFC2617.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_content_length_limit" c:identifier="gst_rtsp_connection_set_content_length_limit" version="1.18">
|
||||
<doc xml:space="preserve">Configure @conn to use the specified Content-Length limit.
|
||||
Both requests and responses are validated. If content-length is
|
||||
exceeded, ENOMEM error will be returned.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="limit" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Content-Length limit</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_http_mode" c:identifier="gst_rtsp_connection_set_http_mode">
|
||||
<doc xml:space="preserve">By setting the HTTP mode to %TRUE the message parsing will support HTTP
|
||||
messages in addition to the RTSP messages. It will also disable the
|
||||
|
@ -828,7 +1059,7 @@ the @conn is connected.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="write" c:identifier="gst_rtsp_connection_write">
|
||||
<method name="write" c:identifier="gst_rtsp_connection_write" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Attempt to write @size bytes of @data to the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be %NULL, in which case this function
|
||||
might block forever.
|
||||
|
@ -858,6 +1089,36 @@ This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="write_usec" c:identifier="gst_rtsp_connection_write_usec" version="1.18">
|
||||
<doc xml:space="preserve">Attempt to write @size bytes of @data to the connected @conn, blocking up to
|
||||
the specified @timeout. @timeout can be 0, in which case this function
|
||||
might block forever.
|
||||
|
||||
This function can be cancelled with gst_rtsp_connection_flush().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">#GST_RTSP_OK on success.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="conn" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPConnection</doc>
|
||||
<type name="RTSPConnection" c:type="GstRTSPConnection*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="data" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the data to write</doc>
|
||||
<type name="guint8" c:type="const guint8*"/>
|
||||
</parameter>
|
||||
<parameter name="size" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the size of @data</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout value or 0</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<function name="accept" c:identifier="gst_rtsp_connection_accept">
|
||||
<doc xml:space="preserve">Accept a new connection on @socket and create a new #GstRTSPConnection for
|
||||
handling communication on new socket.</doc>
|
||||
|
@ -2654,7 +2915,7 @@ UTC times will be converted to nanoseconds since 1900.</doc>
|
|||
<doc xml:space="preserve">no error</doc>
|
||||
</member>
|
||||
<member name="error" value="-1" c:identifier="GST_RTSP_ERROR" glib:nick="error">
|
||||
<doc xml:space="preserve">some unspecified error occured</doc>
|
||||
<doc xml:space="preserve">some unspecified error occurred</doc>
|
||||
</member>
|
||||
<member name="einval" value="-2" c:identifier="GST_RTSP_EINVAL" glib:nick="einval">
|
||||
<doc xml:space="preserve">invalid arguments were provided to a function</doc>
|
||||
|
@ -2666,16 +2927,16 @@ UTC times will be converted to nanoseconds since 1900.</doc>
|
|||
<doc xml:space="preserve">no memory was available for the operation</doc>
|
||||
</member>
|
||||
<member name="eresolv" value="-5" c:identifier="GST_RTSP_ERESOLV" glib:nick="eresolv">
|
||||
<doc xml:space="preserve">a host resolve error occured</doc>
|
||||
<doc xml:space="preserve">a host resolve error occurred</doc>
|
||||
</member>
|
||||
<member name="enotimpl" value="-6" c:identifier="GST_RTSP_ENOTIMPL" glib:nick="enotimpl">
|
||||
<doc xml:space="preserve">function not implemented</doc>
|
||||
</member>
|
||||
<member name="esys" value="-7" c:identifier="GST_RTSP_ESYS" glib:nick="esys">
|
||||
<doc xml:space="preserve">a system error occured, errno contains more details</doc>
|
||||
<doc xml:space="preserve">a system error occurred, errno contains more details</doc>
|
||||
</member>
|
||||
<member name="eparse" value="-8" c:identifier="GST_RTSP_EPARSE" glib:nick="eparse">
|
||||
<doc xml:space="preserve">a parsing error occured</doc>
|
||||
<doc xml:space="preserve">a parsing error occurred</doc>
|
||||
</member>
|
||||
<member name="ewsastart" value="-9" c:identifier="GST_RTSP_EWSASTART" glib:nick="ewsastart">
|
||||
<doc xml:space="preserve">windows networking could not start</doc>
|
||||
|
@ -2687,13 +2948,13 @@ UTC times will be converted to nanoseconds since 1900.</doc>
|
|||
<doc xml:space="preserve">end-of-file was reached</doc>
|
||||
</member>
|
||||
<member name="enet" value="-12" c:identifier="GST_RTSP_ENET" glib:nick="enet">
|
||||
<doc xml:space="preserve">a network problem occured, h_errno contains more details</doc>
|
||||
<doc xml:space="preserve">a network problem occurred, h_errno contains more details</doc>
|
||||
</member>
|
||||
<member name="enotip" value="-13" c:identifier="GST_RTSP_ENOTIP" glib:nick="enotip">
|
||||
<doc xml:space="preserve">the host is not an IP host</doc>
|
||||
</member>
|
||||
<member name="etimeout" value="-14" c:identifier="GST_RTSP_ETIMEOUT" glib:nick="etimeout">
|
||||
<doc xml:space="preserve">a timeout occured</doc>
|
||||
<doc xml:space="preserve">a timeout occurred</doc>
|
||||
</member>
|
||||
<member name="etget" value="-15" c:identifier="GST_RTSP_ETGET" glib:nick="etget">
|
||||
<doc xml:space="preserve">the tunnel GET request has been performed</doc>
|
||||
|
@ -3249,6 +3510,28 @@ g_strfreev() when no longer needed.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_request_uri_with_control" c:identifier="gst_rtsp_url_get_request_uri_with_control">
|
||||
<doc xml:space="preserve">Get a newly allocated string describing the request URI for @url
|
||||
combined with the control path for @control_path</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">a string with the request URI combined with the control path.
|
||||
g_free() after usage.
|
||||
|
||||
Since 1.18</doc>
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="url" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPUrl</doc>
|
||||
<type name="RTSPUrl" c:type="const GstRTSPUrl*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="control_path" transfer-ownership="none">
|
||||
<doc xml:space="preserve">an RTSP aggregate control path</doc>
|
||||
<type name="utf8" c:type="const gchar*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_port" c:identifier="gst_rtsp_url_set_port">
|
||||
<doc xml:space="preserve">Set the port number in @url to @port.</doc>
|
||||
|
||||
|
@ -3493,7 +3776,7 @@ count is zero the watch and associated memory will be destroyed.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="wait_backlog" c:identifier="gst_rtsp_watch_wait_backlog" version="1.4">
|
||||
<method name="wait_backlog" c:identifier="gst_rtsp_watch_wait_backlog" version="1.4" deprecated="1" deprecated-version="1.18">
|
||||
<doc xml:space="preserve">Wait until there is place in the backlog queue, @timeout is reached
|
||||
or @watch is set to flushing.
|
||||
|
||||
|
@ -3518,11 +3801,41 @@ free space in the backlog queue and try again.</doc>
|
|||
<type name="RTSPWatch" c:type="GstRTSPWatch*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GTimeVal timeout</doc>
|
||||
<doc xml:space="preserve">a GTimeVal timeout</doc>
|
||||
<type name="GLib.TimeVal" c:type="GTimeVal*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="wait_backlog_usec" c:identifier="gst_rtsp_watch_wait_backlog_usec" version="1.18">
|
||||
<doc xml:space="preserve">Wait until there is place in the backlog queue, @timeout is reached
|
||||
or @watch is set to flushing.
|
||||
|
||||
If @timeout is 0 this function can block forever. If @timeout
|
||||
contains a valid timeout, this function will return %GST_RTSP_ETIMEOUT
|
||||
after the timeout expired.
|
||||
|
||||
The typically use of this function is when gst_rtsp_watch_write_data
|
||||
returns %GST_RTSP_ENOMEM. The caller then calls this function to wait for
|
||||
free space in the backlog queue and try again.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%GST_RTSP_OK when if there is room in queue.
|
||||
%GST_RTSP_ETIMEOUT when @timeout was reached.
|
||||
%GST_RTSP_EINTR when @watch is flushing
|
||||
%GST_RTSP_EINVAL when called with invalid parameters.</doc>
|
||||
<type name="RTSPResult" c:type="GstRTSPResult"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="watch" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPWatch</doc>
|
||||
<type name="RTSPWatch" c:type="GstRTSPWatch*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="timeout" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a timeout in microseconds</doc>
|
||||
<type name="gint64" c:type="gint64"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="write_data" c:identifier="gst_rtsp_watch_write_data">
|
||||
<doc xml:space="preserve">Write @data using the connection of the @watch. If it cannot be sent
|
||||
immediately, it will be queued for transmission in @watch. The contents of
|
||||
|
|
|
@ -1113,6 +1113,49 @@ no one else overrides it.</doc>
|
|||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
<virtual-method name="adjust_play_mode">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="GstRtsp.RTSPStatusCode" c:type="GstRTSPStatusCode"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="client" transfer-ownership="none">
|
||||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="context" transfer-ownership="none">
|
||||
<type name="RTSPContext" c:type="GstRTSPContext*"/>
|
||||
</parameter>
|
||||
<parameter name="range" transfer-ownership="none">
|
||||
<type name="GstRtsp.RTSPTimeRange" c:type="GstRTSPTimeRange**"/>
|
||||
</parameter>
|
||||
<parameter name="flags" transfer-ownership="none">
|
||||
<type name="Gst.SeekFlags" c:type="GstSeekFlags*"/>
|
||||
</parameter>
|
||||
<parameter name="rate" transfer-ownership="none">
|
||||
<type name="gdouble" c:type="gdouble*"/>
|
||||
</parameter>
|
||||
<parameter name="trickmode_interval" transfer-ownership="none">
|
||||
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
||||
</parameter>
|
||||
<parameter name="enable_rate_control" transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="adjust_play_response">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="GstRtsp.RTSPStatusCode" c:type="GstRTSPStatusCode"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="client" transfer-ownership="none">
|
||||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="context" transfer-ownership="none">
|
||||
<type name="RTSPContext" c:type="GstRTSPContext*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</virtual-method>
|
||||
<virtual-method name="announce_request">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
|
@ -1662,6 +1705,20 @@ The connection object returned remains valid until the client is freed.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_content_length_limit" c:identifier="gst_rtsp_client_get_content_length_limit" version="1.18">
|
||||
<doc xml:space="preserve">Get the Content-Length limit of @client.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">the Content-Length limit.</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="client" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPClient</doc>
|
||||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_mount_points" c:identifier="gst_rtsp_client_get_mount_points">
|
||||
<doc xml:space="preserve">Get the #GstRTSPMountPoints object that @client uses to manage its sessions.</doc>
|
||||
|
||||
|
@ -1690,6 +1747,26 @@ The connection object returned remains valid until the client is freed.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_stream_transport" c:identifier="gst_rtsp_client_get_stream_transport" version="1.18">
|
||||
<doc xml:space="preserve">This is useful when providing a send function through
|
||||
gst_rtsp_client_set_send_func() when doing RTSP over TCP:
|
||||
the send function must call gst_rtsp_stream_transport_message_sent ()
|
||||
on the appropriate transport when data has been received for streaming
|
||||
to continue.</doc>
|
||||
|
||||
<return-value transfer-ownership="none" nullable="1">
|
||||
<doc xml:space="preserve">the #GstRTSPStreamTransport associated with @channel.</doc>
|
||||
<type name="RTSPStreamTransport" c:type="GstRTSPStreamTransport*"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="client" transfer-ownership="none">
|
||||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="channel" transfer-ownership="none">
|
||||
<type name="guint8" c:type="guint8"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_thread_pool" c:identifier="gst_rtsp_client_get_thread_pool">
|
||||
<doc xml:space="preserve">Get the #GstRTSPThreadPool used as the thread pool of @client.</doc>
|
||||
|
||||
|
@ -1821,6 +1898,26 @@ element in the #GList should be unreffed before the list is freed.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_content_length_limit" c:identifier="gst_rtsp_client_set_content_length_limit" version="1.18">
|
||||
<doc xml:space="preserve">Configure @client to use the specified Content-Length limit.
|
||||
|
||||
Define an appropriate request size limit and reject requests exceeding the
|
||||
limit with response status 413 Request Entity Too Large</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="client" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPClient</doc>
|
||||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="limit" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Content-Length limit</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_mount_points" c:identifier="gst_rtsp_client_set_mount_points">
|
||||
<doc xml:space="preserve">Set @mounts as the mount points for @client which it will use to map urls
|
||||
to media streams. These mount points are usually inherited from the server that
|
||||
|
@ -1948,6 +2045,9 @@ that created the client but can be overridden later.</doc>
|
|||
<property name="mount-points" writable="1" transfer-ownership="none">
|
||||
<type name="RTSPMountPoints"/>
|
||||
</property>
|
||||
<property name="post-session-timeout" writable="1" transfer-ownership="none">
|
||||
<type name="gint" c:type="gint"/>
|
||||
</property>
|
||||
<property name="session-pool" writable="1" transfer-ownership="none">
|
||||
<type name="RTSPSessionPool"/>
|
||||
</property>
|
||||
|
@ -2375,6 +2475,53 @@ that created the client but can be overridden later.</doc>
|
|||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="adjust_play_mode">
|
||||
<callback name="adjust_play_mode">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="GstRtsp.RTSPStatusCode" c:type="GstRTSPStatusCode"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="client" transfer-ownership="none">
|
||||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</parameter>
|
||||
<parameter name="context" transfer-ownership="none">
|
||||
<type name="RTSPContext" c:type="GstRTSPContext*"/>
|
||||
</parameter>
|
||||
<parameter name="range" transfer-ownership="none">
|
||||
<type name="GstRtsp.RTSPTimeRange" c:type="GstRTSPTimeRange**"/>
|
||||
</parameter>
|
||||
<parameter name="flags" transfer-ownership="none">
|
||||
<type name="Gst.SeekFlags" c:type="GstSeekFlags*"/>
|
||||
</parameter>
|
||||
<parameter name="rate" transfer-ownership="none">
|
||||
<type name="gdouble" c:type="gdouble*"/>
|
||||
</parameter>
|
||||
<parameter name="trickmode_interval" transfer-ownership="none">
|
||||
<type name="Gst.ClockTime" c:type="GstClockTime*"/>
|
||||
</parameter>
|
||||
<parameter name="enable_rate_control" transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="adjust_play_response">
|
||||
<callback name="adjust_play_response">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="GstRtsp.RTSPStatusCode" c:type="GstRTSPStatusCode"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="client" transfer-ownership="none">
|
||||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</parameter>
|
||||
<parameter name="context" transfer-ownership="none">
|
||||
<type name="RTSPContext" c:type="GstRTSPContext*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</callback>
|
||||
</field>
|
||||
<field name="closed">
|
||||
<callback name="closed">
|
||||
|
||||
|
@ -2820,7 +2967,7 @@ that created the client but can be overridden later.</doc>
|
|||
</callback>
|
||||
</field>
|
||||
<field name="_gst_reserved" readable="0" private="1">
|
||||
<array zero-terminated="0" fixed-size="4">
|
||||
<array zero-terminated="0" fixed-size="2">
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</array>
|
||||
</field>
|
||||
|
@ -3009,7 +3156,7 @@ context can then be received using gst_rtsp_context_get_current().</doc>
|
|||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="ctx" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a ##GstRTSPContext</doc>
|
||||
<doc xml:space="preserve">a #GstRTSPContext</doc>
|
||||
<type name="RTSPContext" c:type="GstRTSPContext*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
|
@ -3646,6 +3793,43 @@ gst_rtsp_media_prepare ().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_rate_control" c:identifier="gst_rtsp_media_get_rate_control" version="1.18">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">whether @media will follow the Rate-Control=no behaviour as specified
|
||||
in the ONVIF replay spec.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="media" transfer-ownership="none">
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_rates" c:identifier="gst_rtsp_media_get_rates" version="1.18">
|
||||
<doc xml:space="preserve">Get the rate and applied_rate of the current segment.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%FALSE if looking up the rate and applied rate failed. Otherwise
|
||||
%TRUE is returned and @rate and @applied_rate are set to the rate and
|
||||
applied_rate of the current segment.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="media" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPMedia
|
||||
@rate (allow-none): the rate of the current segment
|
||||
@applied_rate (allow-none): the applied_rate of the current segment</doc>
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="rate" transfer-ownership="none">
|
||||
<type name="gdouble" c:type="gdouble*"/>
|
||||
</parameter>
|
||||
<parameter name="applied_rate" transfer-ownership="none">
|
||||
<type name="gdouble" c:type="gdouble*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_retransmission_time" c:identifier="gst_rtsp_media_get_retransmission_time">
|
||||
<doc xml:space="preserve">Get the amount of time to store retransmission data.</doc>
|
||||
|
||||
|
@ -3763,6 +3947,19 @@ will listen on @address and @port for client time requests.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="has_completed_sender" c:identifier="gst_rtsp_media_has_completed_sender" version="1.18">
|
||||
<doc xml:space="preserve">See gst_rtsp_stream_is_complete(), gst_rtsp_stream_is_sender().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">whether @media has at least one complete sender stream.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="media" transfer-ownership="none">
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="is_bind_mcast_address" c:identifier="gst_rtsp_media_is_bind_mcast_address" version="1.16">
|
||||
<doc xml:space="preserve">Check if multicast sockets are configured to be bound to multicast addresses.</doc>
|
||||
|
||||
|
@ -3792,6 +3989,18 @@ unpreparing.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="is_receive_only" c:identifier="gst_rtsp_media_is_receive_only" version="1.18">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if @media is receive-only, %FALSE otherwise.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="media" transfer-ownership="none">
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="is_reusable" c:identifier="gst_rtsp_media_is_reusable">
|
||||
<doc xml:space="preserve">Check if the pipeline for @media can be reused after an unprepare.</doc>
|
||||
|
||||
|
@ -3852,6 +4061,27 @@ Use gst_rtsp_media_get_time_provider() to get the network clock.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="lock" c:identifier="gst_rtsp_media_lock" version="1.18">
|
||||
<doc xml:space="preserve">Lock the entire media. This is needed by callers such as rtsp_client to
|
||||
protect the media when it is shared by many clients.
|
||||
The lock prevents that concurrent clients alters the shared media,
|
||||
while one client already is working with it.
|
||||
Typically the lock is taken in external RTSP API calls that uses shared media
|
||||
such as DESCRIBE, SETUP, ANNOUNCE, TEARDOWN, PLAY, PAUSE.
|
||||
|
||||
As best practice take the lock as soon as the function get hold of a shared
|
||||
media object. Release the lock right before the function returns.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="media" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPMedia</doc>
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="n_streams" c:identifier="gst_rtsp_media_n_streams">
|
||||
<doc xml:space="preserve">Get the number of streams in this media.</doc>
|
||||
|
||||
|
@ -3909,10 +4139,9 @@ gst_rtsp_media_prepare().</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="seek_full" c:identifier="gst_rtsp_media_seek_full" version="1.14">
|
||||
<doc xml:space="preserve">Seek the pipeline of @media to @range. @media must be prepared with
|
||||
gst_rtsp_media_prepare(). In order to perform the seek operation,
|
||||
the pipeline must contain all needed transport parts (transport sinks).</doc>
|
||||
<method name="seek_full" c:identifier="gst_rtsp_media_seek_full" version="1.18">
|
||||
<doc xml:space="preserve">Seek the pipeline of @media to @range with the given @flags.
|
||||
@media must be prepared with gst_rtsp_media_prepare().</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE on success.</doc>
|
||||
|
@ -3933,6 +4162,40 @@ the pipeline must contain all needed transport parts (transport sinks).</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="seek_trickmode" c:identifier="gst_rtsp_media_seek_trickmode" version="1.18">
|
||||
<doc xml:space="preserve">Seek the pipeline of @media to @range with the given @flags and @rate,
|
||||
and @trickmode_interval.
|
||||
@media must be prepared with gst_rtsp_media_prepare().
|
||||
In order to perform the seek operation, the pipeline must contain all
|
||||
needed transport parts (transport sinks).</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE on success.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="media" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPMedia</doc>
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="range" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPTimeRange</doc>
|
||||
<type name="GstRtsp.RTSPTimeRange" c:type="GstRTSPTimeRange*"/>
|
||||
</parameter>
|
||||
<parameter name="flags" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The minimal set of #GstSeekFlags to use</doc>
|
||||
<type name="Gst.SeekFlags" c:type="GstSeekFlags"/>
|
||||
</parameter>
|
||||
<parameter name="rate" transfer-ownership="none">
|
||||
<doc xml:space="preserve">the rate to use in the seek</doc>
|
||||
<type name="gdouble" c:type="gdouble"/>
|
||||
</parameter>
|
||||
<parameter name="trickmode_interval" transfer-ownership="none">
|
||||
<doc xml:space="preserve">The trickmode interval to use for KEY_UNITS trick mode</doc>
|
||||
<type name="Gst.ClockTime" c:type="GstClockTime"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="seekable" c:identifier="gst_rtsp_media_seekable" version="1.14">
|
||||
<doc xml:space="preserve">Check if the pipeline for @media seek and up to what point in time,
|
||||
it can seek.</doc>
|
||||
|
@ -4189,6 +4452,22 @@ it is unprepared.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_rate_control" c:identifier="gst_rtsp_media_set_rate_control" version="1.18">
|
||||
<doc xml:space="preserve">Define whether @media will follow the Rate-Control=no behaviour as specified
|
||||
in the ONVIF replay spec.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="media" transfer-ownership="none">
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="enabled" transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_retransmission_time" c:identifier="gst_rtsp_media_set_retransmission_time">
|
||||
<doc xml:space="preserve">Set the amount of time to store retransmission packets.</doc>
|
||||
|
||||
|
@ -4378,12 +4657,25 @@ taken of @pipeline.</doc>
|
|||
<doc xml:space="preserve">a #GstRTSPMedia</doc>
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="pipeline" transfer-ownership="full">
|
||||
<parameter name="pipeline" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstPipeline</doc>
|
||||
<type name="Gst.Pipeline" c:type="GstPipeline*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="unlock" c:identifier="gst_rtsp_media_unlock" version="1.18">
|
||||
<doc xml:space="preserve">Unlock the media.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="media" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPMedia</doc>
|
||||
<type name="RTSPMedia" c:type="GstRTSPMedia*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="unprepare" c:identifier="gst_rtsp_media_unprepare">
|
||||
<doc xml:space="preserve">Unprepare @media. After this call, the media should be prepared again before
|
||||
it can be used again. If the media is set to be non-reusable, a new instance
|
||||
|
@ -5909,7 +6201,7 @@ when a client disconnects without sending TEARDOWN.</doc>
|
|||
|
||||
</record>
|
||||
<class name="RTSPMediaFactoryURI" c:symbol-prefix="rtsp_media_factory_uri" c:type="GstRTSPMediaFactoryURI" parent="RTSPMediaFactory" glib:type-name="GstRTSPMediaFactoryURI" glib:get-type="gst_rtsp_media_factory_uri_get_type" glib:type-struct="RTSPMediaFactoryURIClass">
|
||||
<doc xml:space="preserve">A media factory that creates a pipeline to play and uri.</doc>
|
||||
<doc xml:space="preserve">A media factory that creates a pipeline to play any uri.</doc>
|
||||
|
||||
<constructor name="new" c:identifier="gst_rtsp_media_factory_uri_new">
|
||||
<doc xml:space="preserve">Create a new #GstRTSPMediaFactoryURI instance.</doc>
|
||||
|
@ -6017,7 +6309,10 @@ and called when a message has been sent on the transport.</doc>
|
|||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1" closure="0">
|
||||
<parameter name="trans" transfer-ownership="none">
|
||||
<type name="RTSPStreamTransport" c:type="GstRTSPStreamTransport*"/>
|
||||
</parameter>
|
||||
<parameter name="user_data" transfer-ownership="none" nullable="1" allow-none="1" closure="1">
|
||||
<doc xml:space="preserve">user data</doc>
|
||||
<type name="gpointer" c:type="gpointer"/>
|
||||
</parameter>
|
||||
|
@ -6187,6 +6482,14 @@ g_object_unref() after usage.</doc>
|
|||
</record>
|
||||
<class name="RTSPOnvifClient" c:symbol-prefix="rtsp_onvif_client" c:type="GstRTSPOnvifClient" version="1.14" parent="RTSPClient" glib:type-name="GstRTSPOnvifClient" glib:get-type="gst_rtsp_onvif_client_get_type" glib:type-struct="RTSPOnvifClientClass">
|
||||
|
||||
<constructor name="new" c:identifier="gst_rtsp_onvif_client_new" version="1.18">
|
||||
<doc xml:space="preserve">Create a new #GstRTSPOnvifClient instance.</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">a new #GstRTSPOnvifClient</doc>
|
||||
<type name="RTSPClient" c:type="GstRTSPClient*"/>
|
||||
</return-value>
|
||||
</constructor>
|
||||
<field name="parent">
|
||||
<type name="RTSPClient" c:type="GstRTSPClient"/>
|
||||
</field>
|
||||
|
@ -6370,6 +6673,18 @@ usage.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="has_replay_support" c:identifier="gst_rtsp_onvif_media_factory_has_replay_support" version="1.18">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if ONVIF replay is supported by the media factory.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="factory" transfer-ownership="none">
|
||||
<type name="RTSPOnvifMediaFactory" c:type="GstRTSPOnvifMediaFactory*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_backchannel_bandwidth" c:identifier="gst_rtsp_onvif_media_factory_set_backchannel_bandwidth" version="1.14">
|
||||
<doc xml:space="preserve">Set the configured/supported bandwidth of the ONVIF backchannel pipeline in
|
||||
bits per second.</doc>
|
||||
|
@ -6420,6 +6735,21 @@ prepare.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_replay_support" c:identifier="gst_rtsp_onvif_media_factory_set_replay_support" version="1.18">
|
||||
<doc xml:space="preserve">Set to %TRUE if ONVIF replay is supported by the media factory.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="factory" transfer-ownership="none">
|
||||
<type name="RTSPOnvifMediaFactory" c:type="GstRTSPOnvifMediaFactory*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="has_replay_support" transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<field name="parent">
|
||||
<type name="RTSPMediaFactory" c:type="GstRTSPMediaFactory"/>
|
||||
</field>
|
||||
|
@ -6974,6 +7304,20 @@ usage.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_content_length_limit" c:identifier="gst_rtsp_server_get_content_length_limit" version="1.18">
|
||||
<doc xml:space="preserve">Get the Content-Length limit of @server.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">the Content-Length limit.</doc>
|
||||
<type name="guint" c:type="guint"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="server" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPServer</doc>
|
||||
<type name="RTSPServer" c:type="GstRTSPServer*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_mount_points" c:identifier="gst_rtsp_server_get_mount_points">
|
||||
<doc xml:space="preserve">Get the #GstRTSPMountPoints used as the mount points of @server.</doc>
|
||||
|
||||
|
@ -7089,6 +7433,24 @@ This function must be called before the server is bound.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_content_length_limit" c:identifier="gst_rtsp_server_set_content_length_limit" version="1.18">
|
||||
<doc xml:space="preserve">Define an appropriate request size limit and reject requests exceeding the
|
||||
limit.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="server" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPServer
|
||||
Configure @server to use the specified Content-Length limit.</doc>
|
||||
<type name="RTSPServer" c:type="GstRTSPServer*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="limit" transfer-ownership="none">
|
||||
<type name="guint" c:type="guint"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_mount_points" c:identifier="gst_rtsp_server_set_mount_points">
|
||||
<doc xml:space="preserve">configure @mounts to be used as the mount points of @server.</doc>
|
||||
|
||||
|
@ -7207,6 +7569,9 @@ that the HTTP server read from the socket while parsing the HTTP header.</doc>
|
|||
<property name="bound-port" transfer-ownership="none">
|
||||
<type name="gint" c:type="gint"/>
|
||||
</property>
|
||||
<property name="content-length-limit" writable="1" transfer-ownership="none">
|
||||
<type name="guint" c:type="guint"/>
|
||||
</property>
|
||||
<property name="mount-points" writable="1" transfer-ownership="none">
|
||||
<type name="RTSPMountPoints"/>
|
||||
</property>
|
||||
|
@ -7614,6 +7979,9 @@ cleaned up when there is no activity for @timeout seconds.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<property name="extra-timeout" writable="1" transfer-ownership="none">
|
||||
<type name="guint" c:type="guint"/>
|
||||
</property>
|
||||
<property name="sessionid" writable="1" construct-only="1" transfer-ownership="none">
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</property>
|
||||
|
@ -8024,8 +8392,7 @@ what happens to the session. @func will be called with the session pool
|
|||
locked so no further actions on @pool can be performed from @func.
|
||||
|
||||
If @func returns #GST_RTSP_FILTER_REMOVE, the session will be set to the
|
||||
expired state with gst_rtsp_session_set_expired() and removed from
|
||||
@pool.
|
||||
expired state and removed from @pool.
|
||||
|
||||
If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @pool.
|
||||
|
||||
|
@ -8344,7 +8711,8 @@ allocated.</doc>
|
|||
</method>
|
||||
<method name="add_transport" c:identifier="gst_rtsp_stream_add_transport">
|
||||
<doc xml:space="preserve">Add the transport in @trans to @stream. The media of @stream will
|
||||
then also be send to the values configured in @trans.
|
||||
then also be send to the values configured in @trans. Adding the
|
||||
same transport twice will not add it a second time.
|
||||
|
||||
@stream must be joined to a bin.
|
||||
|
||||
|
@ -8656,6 +9024,41 @@ g_free() after usage.</doc>
|
|||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_rate_control" c:identifier="gst_rtsp_stream_get_rate_control" version="1.18">
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">whether @stream will follow the Rate-Control=no behaviour as specified
|
||||
in the ONVIF replay spec.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="stream" transfer-ownership="none">
|
||||
<type name="RTSPStream" c:type="GstRTSPStream*"/>
|
||||
</instance-parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_rates" c:identifier="gst_rtsp_stream_get_rates" version="1.18">
|
||||
<doc xml:space="preserve">Retrieve the current rate and/or applied_rate.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">%TRUE if rate and/or applied_rate could be determined.</doc>
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="stream" transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstRTSPStream</doc>
|
||||
<type name="RTSPStream" c:type="GstRTSPStream*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="rate" transfer-ownership="none" nullable="1" allow-none="1">
|
||||
<doc xml:space="preserve">the configured rate</doc>
|
||||
<type name="gdouble" c:type="gdouble*"/>
|
||||
</parameter>
|
||||
<parameter name="applied_rate" transfer-ownership="none" nullable="1" allow-none="1">
|
||||
<doc xml:space="preserve">the configured applied_rate</doc>
|
||||
<type name="gdouble" c:type="gdouble*"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="get_retransmission_pt" c:identifier="gst_rtsp_stream_get_retransmission_pt">
|
||||
<doc xml:space="preserve">Get the payload-type used for retransmission of this stream</doc>
|
||||
|
||||
|
@ -9591,6 +9994,22 @@ an RTSP connection.</doc>
|
|||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_rate_control" c:identifier="gst_rtsp_stream_set_rate_control" version="1.18">
|
||||
<doc xml:space="preserve">Define whether @stream will follow the Rate-Control=no behaviour as specified
|
||||
in the ONVIF replay spec.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<type name="none" c:type="void"/>
|
||||
</return-value>
|
||||
<parameters>
|
||||
<instance-parameter name="stream" transfer-ownership="none">
|
||||
<type name="RTSPStream" c:type="GstRTSPStream*"/>
|
||||
</instance-parameter>
|
||||
<parameter name="enabled" transfer-ownership="none">
|
||||
<type name="gboolean" c:type="gboolean"/>
|
||||
</parameter>
|
||||
</parameters>
|
||||
</method>
|
||||
<method name="set_retransmission_pt" c:identifier="gst_rtsp_stream_set_retransmission_pt">
|
||||
<doc xml:space="preserve">Set the payload type (pt) for retransmission of this stream.</doc>
|
||||
|
||||
|
@ -11203,6 +11622,10 @@ port pair in multicast. No response is sent when the check returns
|
|||
</parameter>
|
||||
</parameters>
|
||||
</function-macro>
|
||||
<constant name="RTSP_ONVIF_REPLAY_REQUIREMENT" value="onvif-replay" c:type="GST_RTSP_ONVIF_REPLAY_REQUIREMENT">
|
||||
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</constant>
|
||||
<function-macro name="RTSP_ONVIF_SERVER" c:identifier="GST_RTSP_ONVIF_SERVER" introspectable="0">
|
||||
|
||||
<parameters>
|
||||
|
|
|
@ -379,7 +379,7 @@ in NTP-UTC format.</doc>
|
|||
</parameters>
|
||||
</method>
|
||||
<method name="find_payload" c:identifier="gst_mikey_message_find_payload" version="1.4">
|
||||
<doc xml:space="preserve">Find the @nth occurence of the payload with @type in @msg.</doc>
|
||||
<doc xml:space="preserve">Find the @nth occurrence of the payload with @type in @msg.</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">the @nth #GstMIKEYPayload of @type.</doc>
|
||||
|
@ -1218,7 +1218,7 @@ specific security protocol</doc>
|
|||
</field>
|
||||
</record>
|
||||
<record name="MIKEYPayloadSPParam" c:type="GstMIKEYPayloadSPParam">
|
||||
<doc xml:space="preserve">A Type/Length/Value field for security paramaters</doc>
|
||||
<doc xml:space="preserve">A Type/Length/Value field for security parameters</doc>
|
||||
|
||||
<field name="type" writable="1">
|
||||
<doc xml:space="preserve">specifies the type of the parameter</doc>
|
||||
|
@ -1280,7 +1280,7 @@ specific security protocol</doc>
|
|||
<doc xml:space="preserve">Cert hash payload</doc>
|
||||
</member>
|
||||
<member name="v" value="9" c:identifier="GST_MIKEY_PT_V">
|
||||
<doc xml:space="preserve">Verfication message payload</doc>
|
||||
<doc xml:space="preserve">Verification message payload</doc>
|
||||
</member>
|
||||
<member name="sp" value="10" c:identifier="GST_MIKEY_PT_SP">
|
||||
<doc xml:space="preserve">Security Policy payload</doc>
|
||||
|
@ -1935,7 +1935,9 @@ a=rtpmap:(payload) (encoding_name)/(clock_rate)[/(encoding_params)]
|
|||
|
||||
a=framesize:(payload) (width)-(height)
|
||||
|
||||
a=fmtp:(payload) (param)[=(value)];...</doc>
|
||||
a=fmtp:(payload) (param)[=(value)];...
|
||||
|
||||
Note that the extmap attribute is set only by gst_sdp_media_attributes_to_caps().</doc>
|
||||
|
||||
<return-value transfer-ownership="full">
|
||||
<doc xml:space="preserve">a #GstCaps, or %NULL if an error happened</doc>
|
||||
|
@ -2499,7 +2501,9 @@ a=framesize:(payload) (width)-(height)
|
|||
|
||||
a=fmtp:(payload) (param)[=(value)];...
|
||||
|
||||
a=rtcp-fb:(payload) (param1) [param2]...</doc>
|
||||
a=rtcp-fb:(payload) (param1) [param2]...
|
||||
|
||||
a=extmap:(id)[/direction] (extensionname) (extensionattributes)</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstSDPResult.</doc>
|
||||
|
@ -4158,7 +4162,9 @@ a=framesize:(payload) (width)-(height)
|
|||
|
||||
a=fmtp:(payload) (param)[=(value)];...
|
||||
|
||||
a=rtcp-fb:(payload) (param1) [param2]...</doc>
|
||||
a=rtcp-fb:(payload) (param1) [param2]...
|
||||
|
||||
a=extmap:(id)[/direction] (extensionname) (extensionattributes)</doc>
|
||||
|
||||
<return-value transfer-ownership="none">
|
||||
<doc xml:space="preserve">a #GstSDPResult.</doc>
|
||||
|
|
|
@ -43,6 +43,16 @@ and/or use gtk-doc annotations. -->
|
|||
</parameter>
|
||||
</parameters>
|
||||
</function-macro>
|
||||
<constant name="TAG_ACOUSTID_FINGERPRINT" value="chromaprint-fingerprint" c:type="GST_TAG_ACOUSTID_FINGERPRINT" version="1.18">
|
||||
<doc xml:space="preserve">AcoustID Fingerprint (Chromaprint)</doc>
|
||||
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</constant>
|
||||
<constant name="TAG_ACOUSTID_ID" value="acoustid-id" c:type="GST_TAG_ACOUSTID_ID" version="1.18">
|
||||
<doc xml:space="preserve">AcoustID Identifier</doc>
|
||||
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</constant>
|
||||
<constant name="TAG_CAPTURING_CONTRAST" value="capturing-contrast" c:type="GST_TAG_CAPTURING_CONTRAST">
|
||||
<doc xml:space="preserve">Direction of contrast processing applied when capturing an image. (string)
|
||||
|
||||
|
@ -311,6 +321,16 @@ keys.</doc>
|
|||
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</constant>
|
||||
<constant name="TAG_MUSICBRAINZ_RELEASEGROUPID" value="musicbrainz-releasegroupid" c:type="GST_TAG_MUSICBRAINZ_RELEASEGROUPID" version="1.18">
|
||||
<doc xml:space="preserve">MusicBrainz Release Group ID</doc>
|
||||
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</constant>
|
||||
<constant name="TAG_MUSICBRAINZ_RELEASETRACKID" value="musicbrainz-releasetrackid" c:type="GST_TAG_MUSICBRAINZ_RELEASETRACKID" version="1.18">
|
||||
<doc xml:space="preserve">MusicBrainz Release Track ID</doc>
|
||||
|
||||
<type name="utf8" c:type="gchar*"/>
|
||||
</constant>
|
||||
<constant name="TAG_MUSICBRAINZ_TRACKID" value="musicbrainz-trackid" c:type="GST_TAG_MUSICBRAINZ_TRACKID">
|
||||
<doc xml:space="preserve">MusicBrainz track ID</doc>
|
||||
|
||||
|
@ -1062,7 +1082,7 @@ code (both are accepted for convenience).
|
|||
|
||||
The "bibliographic" code is derived from the English name of the language
|
||||
(e.g. "ger" for German instead of "de" or "deu"). In most scenarios, the
|
||||
"terminological" codes are prefered.
|
||||
"terminological" codes are preferred.
|
||||
|
||||
Language codes are case-sensitive and expected to be lower case.</doc>
|
||||
|
||||
|
@ -1086,7 +1106,7 @@ code (both are accepted for convenience).
|
|||
|
||||
The "terminological" code is derived from the local name of the language
|
||||
(e.g. "deu" for German instead of "ger"). In most scenarios, the
|
||||
"terminological" codes are prefered over the "bibliographic" ones.
|
||||
"terminological" codes are preferred over the "bibliographic" ones.
|
||||
|
||||
Language codes are case-sensitive and expected to be lower case.</doc>
|
||||
|
||||
|
@ -1296,7 +1316,7 @@ rather than the image itself.
|
|||
In GStreamer, image tags are #GstSample<!-- -->s containing the raw image
|
||||
data, with the sample caps describing the content type of the image
|
||||
(e.g. image/jpeg, image/png, text/uri-list). The sample info may contain
|
||||
an additional 'image-type' field of #GST_TYPE_TAG_IMAGE_TYPE to describe
|
||||
an additional 'image-type' field of #GstTagImageType to describe
|
||||
the type of image (front cover, back cover etc.). #GST_TAG_PREVIEW_IMAGE
|
||||
tags should not carry an image type, their type is already indicated via
|
||||
the special tag name.
|
||||
|
|
File diff suppressed because it is too large
Load diff
|
@ -200,17 +200,17 @@ for more information.</doc>
|
|||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCDTLSSetup" glib:type-name="GstWebRTCDTLSSetup" glib:get-type="gst_webrtc_dtls_setup_get_type" c:type="GstWebRTCDTLSSetup">
|
||||
<doc xml:space="preserve">GST_WEBRTC_DTLS_SETUP_NONE: none
|
||||
GST_WEBRTC_DTLS_SETUP_ACTPASS: actpass
|
||||
GST_WEBRTC_DTLS_SETUP_ACTIVE: sendonly
|
||||
GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
|
||||
<member name="none" value="0" c:identifier="GST_WEBRTC_DTLS_SETUP_NONE" glib:nick="none">
|
||||
<doc xml:space="preserve">none</doc>
|
||||
</member>
|
||||
<member name="actpass" value="1" c:identifier="GST_WEBRTC_DTLS_SETUP_ACTPASS" glib:nick="actpass">
|
||||
<doc xml:space="preserve">actpass</doc>
|
||||
</member>
|
||||
<member name="active" value="2" c:identifier="GST_WEBRTC_DTLS_SETUP_ACTIVE" glib:nick="active">
|
||||
<doc xml:space="preserve">sendonly</doc>
|
||||
</member>
|
||||
<member name="passive" value="3" c:identifier="GST_WEBRTC_DTLS_SETUP_PASSIVE" glib:nick="passive">
|
||||
<doc xml:space="preserve">recvonly</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<class name="WebRTCDTLSTransport" c:symbol-prefix="webrtc_dtls_transport" c:type="GstWebRTCDTLSTransport" parent="Gst.Object" glib:type-name="GstWebRTCDTLSTransport" glib:get-type="gst_webrtc_dtls_transport_get_type" glib:type-struct="WebRTCDTLSTransportClass">
|
||||
|
@ -306,20 +306,20 @@ GST_WEBRTC_DTLS_SETUP_PASSIVE: recvonly</doc>
|
|||
</field>
|
||||
</record>
|
||||
<enumeration name="WebRTCDTLSTransportState" glib:type-name="GstWebRTCDTLSTransportState" glib:get-type="gst_webrtc_dtls_transport_state_get_type" c:type="GstWebRTCDTLSTransportState">
|
||||
<doc xml:space="preserve">GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW: new
|
||||
GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED: closed
|
||||
GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED: failed
|
||||
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING: connecting
|
||||
GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED: connected</doc>
|
||||
<member name="new" value="0" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_NEW" glib:nick="new">
|
||||
<doc xml:space="preserve">new</doc>
|
||||
</member>
|
||||
<member name="closed" value="1" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CLOSED" glib:nick="closed">
|
||||
<doc xml:space="preserve">closed</doc>
|
||||
</member>
|
||||
<member name="failed" value="2" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_FAILED" glib:nick="failed">
|
||||
<doc xml:space="preserve">failed</doc>
|
||||
</member>
|
||||
<member name="connecting" value="3" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTING" glib:nick="connecting">
|
||||
<doc xml:space="preserve">connecting</doc>
|
||||
</member>
|
||||
<member name="connected" value="4" c:identifier="GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED" glib:nick="connected">
|
||||
<doc xml:space="preserve">connected</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCDataChannelState" version="1.16" glib:type-name="GstWebRTCDataChannelState" glib:get-type="gst_webrtc_data_channel_state_get_type" c:type="GstWebRTCDataChannelState">
|
||||
|
@ -328,7 +328,7 @@ GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING: connection
|
|||
GST_WEBRTC_DATA_CHANNEL_STATE_OPEN: open
|
||||
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING: closing
|
||||
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED: closed
|
||||
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate</ulink></doc>
|
||||
See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate></doc>
|
||||
<member name="new" value="0" c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_NEW" glib:nick="new">
|
||||
</member>
|
||||
<member name="connecting" value="1" c:identifier="GST_WEBRTC_DATA_CHANNEL_STATE_CONNECTING" glib:nick="connecting">
|
||||
|
@ -349,55 +349,55 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate">h
|
|||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCICEComponent" glib:type-name="GstWebRTCICEComponent" glib:get-type="gst_webrtc_ice_component_get_type" c:type="GstWebRTCICEComponent">
|
||||
<doc xml:space="preserve">GST_WEBRTC_ICE_COMPONENT_RTP,
|
||||
GST_WEBRTC_ICE_COMPONENT_RTCP,</doc>
|
||||
<member name="rtp" value="0" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTP" glib:nick="rtp">
|
||||
<doc xml:space="preserve">RTP component</doc>
|
||||
</member>
|
||||
<member name="rtcp" value="1" c:identifier="GST_WEBRTC_ICE_COMPONENT_RTCP" glib:nick="rtcp">
|
||||
<doc xml:space="preserve">RTCP component</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCICEConnectionState" glib:type-name="GstWebRTCICEConnectionState" glib:get-type="gst_webrtc_ice_connection_state_get_type" c:type="GstWebRTCICEConnectionState">
|
||||
<doc xml:space="preserve">GST_WEBRTC_ICE_CONNECTION_STATE_NEW: new
|
||||
GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING: checking
|
||||
GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED: connected
|
||||
GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED: completed
|
||||
GST_WEBRTC_ICE_CONNECTION_STATE_FAILED: failed
|
||||
GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED: disconnected
|
||||
GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED: closed
|
||||
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate</ulink></doc>
|
||||
<doc xml:space="preserve">See <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate></doc>
|
||||
<member name="new" value="0" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_NEW" glib:nick="new">
|
||||
<doc xml:space="preserve">new</doc>
|
||||
</member>
|
||||
<member name="checking" value="1" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CHECKING" glib:nick="checking">
|
||||
<doc xml:space="preserve">checking</doc>
|
||||
</member>
|
||||
<member name="connected" value="2" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED" glib:nick="connected">
|
||||
<doc xml:space="preserve">connected</doc>
|
||||
</member>
|
||||
<member name="completed" value="3" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED" glib:nick="completed">
|
||||
<doc xml:space="preserve">completed</doc>
|
||||
</member>
|
||||
<member name="failed" value="4" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_FAILED" glib:nick="failed">
|
||||
<doc xml:space="preserve">failed</doc>
|
||||
</member>
|
||||
<member name="disconnected" value="5" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED" glib:nick="disconnected">
|
||||
<doc xml:space="preserve">disconnected</doc>
|
||||
</member>
|
||||
<member name="closed" value="6" c:identifier="GST_WEBRTC_ICE_CONNECTION_STATE_CLOSED" glib:nick="closed">
|
||||
<doc xml:space="preserve">closed</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCICEGatheringState" glib:type-name="GstWebRTCICEGatheringState" glib:get-type="gst_webrtc_ice_gathering_state_get_type" c:type="GstWebRTCICEGatheringState">
|
||||
<doc xml:space="preserve">GST_WEBRTC_ICE_GATHERING_STATE_NEW: new
|
||||
GST_WEBRTC_ICE_GATHERING_STATE_GATHERING: gathering
|
||||
GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE: complete
|
||||
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate">http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate</ulink></doc>
|
||||
<doc xml:space="preserve">See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate></doc>
|
||||
<member name="new" value="0" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_NEW" glib:nick="new">
|
||||
<doc xml:space="preserve">new</doc>
|
||||
</member>
|
||||
<member name="gathering" value="1" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_GATHERING" glib:nick="gathering">
|
||||
<doc xml:space="preserve">gathering</doc>
|
||||
</member>
|
||||
<member name="complete" value="2" c:identifier="GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE" glib:nick="complete">
|
||||
<doc xml:space="preserve">complete</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCICERole" glib:type-name="GstWebRTCICERole" glib:get-type="gst_webrtc_ice_role_get_type" c:type="GstWebRTCICERole">
|
||||
<doc xml:space="preserve">GST_WEBRTC_ICE_ROLE_CONTROLLED: controlled
|
||||
GST_WEBRTC_ICE_ROLE_CONTROLLING: controlling</doc>
|
||||
<member name="controlled" value="0" c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLED" glib:nick="controlled">
|
||||
<doc xml:space="preserve">controlled</doc>
|
||||
</member>
|
||||
<member name="controlling" value="1" c:identifier="GST_WEBRTC_ICE_ROLE_CONTROLLING" glib:nick="controlling">
|
||||
<doc xml:space="preserve">controlling</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<class name="WebRTCICETransport" c:symbol-prefix="webrtc_ice_transport" c:type="GstWebRTCICETransport" parent="Gst.Object" abstract="1" glib:type-name="GstWebRTCICETransport" glib:get-type="gst_webrtc_ice_transport_get_type" glib:type-struct="WebRTCICETransportClass">
|
||||
|
@ -558,24 +558,24 @@ for more information.</doc>
|
|||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCPeerConnectionState" glib:type-name="GstWebRTCPeerConnectionState" glib:get-type="gst_webrtc_peer_connection_state_get_type" c:type="GstWebRTCPeerConnectionState">
|
||||
<doc xml:space="preserve">GST_WEBRTC_PEER_CONNECTION_STATE_NEW: new
|
||||
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING: connecting
|
||||
GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED: connected
|
||||
GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED: disconnected
|
||||
GST_WEBRTC_PEER_CONNECTION_STATE_FAILED: failed
|
||||
GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED: closed
|
||||
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate">http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate</ulink></doc>
|
||||
<doc xml:space="preserve">See <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate></doc>
|
||||
<member name="new" value="0" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_NEW" glib:nick="new">
|
||||
<doc xml:space="preserve">new</doc>
|
||||
</member>
|
||||
<member name="connecting" value="1" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTING" glib:nick="connecting">
|
||||
<doc xml:space="preserve">connecting</doc>
|
||||
</member>
|
||||
<member name="connected" value="2" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CONNECTED" glib:nick="connected">
|
||||
<doc xml:space="preserve">connected</doc>
|
||||
</member>
|
||||
<member name="disconnected" value="3" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED" glib:nick="disconnected">
|
||||
<doc xml:space="preserve">disconnected</doc>
|
||||
</member>
|
||||
<member name="failed" value="4" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_FAILED" glib:nick="failed">
|
||||
<doc xml:space="preserve">failed</doc>
|
||||
</member>
|
||||
<member name="closed" value="5" c:identifier="GST_WEBRTC_PEER_CONNECTION_STATE_CLOSED" glib:nick="closed">
|
||||
<doc xml:space="preserve">closed</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCPriorityType" version="1.16" glib:type-name="GstWebRTCPriorityType" glib:get-type="gst_webrtc_priority_type_get_type" c:type="GstWebRTCPriorityType">
|
||||
|
@ -583,7 +583,7 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate"&g
|
|||
GST_WEBRTC_PRIORITY_TYPE_LOW: low
|
||||
GST_WEBRTC_PRIORITY_TYPE_MEDIUM: medium
|
||||
GST_WEBRTC_PRIORITY_TYPE_HIGH: high
|
||||
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype</ulink></doc>
|
||||
See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype></doc>
|
||||
<member name="very_low" value="1" c:identifier="GST_WEBRTC_PRIORITY_TYPE_VERY_LOW" glib:nick="very-low">
|
||||
</member>
|
||||
<member name="low" value="2" c:identifier="GST_WEBRTC_PRIORITY_TYPE_LOW" glib:nick="low">
|
||||
|
@ -724,6 +724,10 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http:
|
|||
</record>
|
||||
<class name="WebRTCRTPTransceiver" c:symbol-prefix="webrtc_rtp_transceiver" c:type="GstWebRTCRTPTransceiver" parent="Gst.Object" abstract="1" glib:type-name="GstWebRTCRTPTransceiver" glib:get-type="gst_webrtc_rtp_transceiver_get_type" glib:type-struct="WebRTCRTPTransceiverClass">
|
||||
|
||||
<property name="direction" version="1.18" writable="1" transfer-ownership="none">
|
||||
<doc xml:space="preserve">Direction of the transceiver.</doc>
|
||||
<type name="WebRTCRTPTransceiverDirection"/>
|
||||
</property>
|
||||
<property name="mlineindex" writable="1" construct-only="1" transfer-ownership="none">
|
||||
<type name="guint" c:type="guint"/>
|
||||
</property>
|
||||
|
@ -779,14 +783,19 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http:
|
|||
</record>
|
||||
<enumeration name="WebRTCRTPTransceiverDirection" glib:type-name="GstWebRTCRTPTransceiverDirection" glib:get-type="gst_webrtc_rtp_transceiver_direction_get_type" c:type="GstWebRTCRTPTransceiverDirection">
|
||||
<member name="none" value="0" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE" glib:nick="none">
|
||||
<doc xml:space="preserve">none</doc>
|
||||
</member>
|
||||
<member name="inactive" value="1" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE" glib:nick="inactive">
|
||||
<doc xml:space="preserve">inactive</doc>
|
||||
</member>
|
||||
<member name="sendonly" value="2" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY" glib:nick="sendonly">
|
||||
<doc xml:space="preserve">sendonly</doc>
|
||||
</member>
|
||||
<member name="recvonly" value="3" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY" glib:nick="recvonly">
|
||||
<doc xml:space="preserve">recvonly</doc>
|
||||
</member>
|
||||
<member name="sendrecv" value="4" c:identifier="GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV" glib:nick="sendrecv">
|
||||
<doc xml:space="preserve">sendrecv</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCSCTPTransportState" version="1.16" glib:type-name="GstWebRTCSCTPTransportState" glib:get-type="gst_webrtc_sctp_transport_state_get_type" c:type="GstWebRTCSCTPTransportState">
|
||||
|
@ -794,7 +803,7 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype">http:
|
|||
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING: connecting
|
||||
GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED: connected
|
||||
GST_WEBRTC_SCTP_TRANSPORT_STATE_CLOSED: closed
|
||||
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate</ulink></doc>
|
||||
See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate></doc>
|
||||
<member name="new" value="0" c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_NEW" glib:nick="new">
|
||||
</member>
|
||||
<member name="connecting" value="1" c:identifier="GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTING" glib:nick="connecting">
|
||||
|
@ -805,18 +814,18 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate">
|
|||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCSDPType" glib:type-name="GstWebRTCSDPType" glib:get-type="gst_webrtc_sdp_type_get_type" c:type="GstWebRTCSDPType">
|
||||
<doc xml:space="preserve">GST_WEBRTC_SDP_TYPE_OFFER: offer
|
||||
GST_WEBRTC_SDP_TYPE_PRANSWER: pranswer
|
||||
GST_WEBRTC_SDP_TYPE_ANSWER: answer
|
||||
GST_WEBRTC_SDP_TYPE_ROLLBACK: rollback
|
||||
See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.github.io/webrtc-pc/#rtcsdptype</ulink></doc>
|
||||
<doc xml:space="preserve">See <http://w3c.github.io/webrtc-pc/#rtcsdptype></doc>
|
||||
<member name="offer" value="1" c:identifier="GST_WEBRTC_SDP_TYPE_OFFER" glib:nick="offer">
|
||||
<doc xml:space="preserve">offer</doc>
|
||||
</member>
|
||||
<member name="pranswer" value="2" c:identifier="GST_WEBRTC_SDP_TYPE_PRANSWER" glib:nick="pranswer">
|
||||
<doc xml:space="preserve">pranswer</doc>
|
||||
</member>
|
||||
<member name="answer" value="3" c:identifier="GST_WEBRTC_SDP_TYPE_ANSWER" glib:nick="answer">
|
||||
<doc xml:space="preserve">answer</doc>
|
||||
</member>
|
||||
<member name="rollback" value="4" c:identifier="GST_WEBRTC_SDP_TYPE_ROLLBACK" glib:nick="rollback">
|
||||
<doc xml:space="preserve">rollback</doc>
|
||||
</member>
|
||||
<function name="to_string" c:identifier="gst_webrtc_sdp_type_to_string">
|
||||
|
||||
|
@ -834,7 +843,7 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.git
|
|||
</function>
|
||||
</enumeration>
|
||||
<record name="WebRTCSessionDescription" c:type="GstWebRTCSessionDescription" glib:type-name="GstWebRTCSessionDescription" glib:get-type="gst_webrtc_session_description_get_type" c:symbol-prefix="webrtc_session_description">
|
||||
<doc xml:space="preserve">See <ulink url="https://www.w3.org/TR/webrtc/#rtcsessiondescription-class">https://www.w3.org/TR/webrtc/#rtcsessiondescription-class</ulink></doc>
|
||||
<doc xml:space="preserve">See <https://www.w3.org/TR/webrtc/#rtcsessiondescription-class></doc>
|
||||
|
||||
<field name="type" writable="1">
|
||||
<doc xml:space="preserve">the #GstWebRTCSDPType of the description</doc>
|
||||
|
@ -890,68 +899,68 @@ See <ulink url="http://w3c.github.io/webrtc-pc/#rtcsdptype">http://w3c.git
|
|||
</method>
|
||||
</record>
|
||||
<enumeration name="WebRTCSignalingState" glib:type-name="GstWebRTCSignalingState" glib:get-type="gst_webrtc_signaling_state_get_type" c:type="GstWebRTCSignalingState">
|
||||
<doc xml:space="preserve">GST_WEBRTC_SIGNALING_STATE_STABLE: stable
|
||||
GST_WEBRTC_SIGNALING_STATE_CLOSED: closed
|
||||
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER: have-local-offer
|
||||
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER: have-remote-offer
|
||||
GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER: have-local-pranswer
|
||||
GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER: have-remote-pranswer
|
||||
See <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate">http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate</ulink></doc>
|
||||
<doc xml:space="preserve">See <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate></doc>
|
||||
<member name="stable" value="0" c:identifier="GST_WEBRTC_SIGNALING_STATE_STABLE" glib:nick="stable">
|
||||
<doc xml:space="preserve">stable</doc>
|
||||
</member>
|
||||
<member name="closed" value="1" c:identifier="GST_WEBRTC_SIGNALING_STATE_CLOSED" glib:nick="closed">
|
||||
<doc xml:space="preserve">closed</doc>
|
||||
</member>
|
||||
<member name="have_local_offer" value="2" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER" glib:nick="have-local-offer">
|
||||
<doc xml:space="preserve">have-local-offer</doc>
|
||||
</member>
|
||||
<member name="have_remote_offer" value="3" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER" glib:nick="have-remote-offer">
|
||||
<doc xml:space="preserve">have-remote-offer</doc>
|
||||
</member>
|
||||
<member name="have_local_pranswer" value="4" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER" glib:nick="have-local-pranswer">
|
||||
<doc xml:space="preserve">have-local-pranswer</doc>
|
||||
</member>
|
||||
<member name="have_remote_pranswer" value="5" c:identifier="GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER" glib:nick="have-remote-pranswer">
|
||||
<doc xml:space="preserve">have-remote-pranswer</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<enumeration name="WebRTCStatsType" glib:type-name="GstWebRTCStatsType" glib:get-type="gst_webrtc_stats_type_get_type" c:type="GstWebRTCStatsType">
|
||||
<doc xml:space="preserve">GST_WEBRTC_STATS_CODEC: codec
|
||||
GST_WEBRTC_STATS_INBOUND_RTP: inbound-rtp
|
||||
GST_WEBRTC_STATS_OUTBOUND_RTP: outbound-rtp
|
||||
GST_WEBRTC_STATS_REMOTE_INBOUND_RTP: remote-inbound-rtp
|
||||
GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP: remote-outbound-rtp
|
||||
GST_WEBRTC_STATS_CSRC: csrc
|
||||
GST_WEBRTC_STATS_PEER_CONNECTION: peer-connectiion
|
||||
GST_WEBRTC_STATS_DATA_CHANNEL: data-channel
|
||||
GST_WEBRTC_STATS_STREAM: stream
|
||||
GST_WEBRTC_STATS_TRANSPORT: transport
|
||||
GST_WEBRTC_STATS_CANDIDATE_PAIR: candidate-pair
|
||||
GST_WEBRTC_STATS_LOCAL_CANDIDATE: local-candidate
|
||||
GST_WEBRTC_STATS_REMOTE_CANDIDATE: remote-candidate
|
||||
GST_WEBRTC_STATS_CERTIFICATE: certificate</doc>
|
||||
<member name="codec" value="1" c:identifier="GST_WEBRTC_STATS_CODEC" glib:nick="codec">
|
||||
<doc xml:space="preserve">codec</doc>
|
||||
</member>
|
||||
<member name="inbound_rtp" value="2" c:identifier="GST_WEBRTC_STATS_INBOUND_RTP" glib:nick="inbound-rtp">
|
||||
<doc xml:space="preserve">inbound-rtp</doc>
|
||||
</member>
|
||||
<member name="outbound_rtp" value="3" c:identifier="GST_WEBRTC_STATS_OUTBOUND_RTP" glib:nick="outbound-rtp">
|
||||
<doc xml:space="preserve">outbound-rtp</doc>
|
||||
</member>
|
||||
<member name="remote_inbound_rtp" value="4" c:identifier="GST_WEBRTC_STATS_REMOTE_INBOUND_RTP" glib:nick="remote-inbound-rtp">
|
||||
<doc xml:space="preserve">remote-inbound-rtp</doc>
|
||||
</member>
|
||||
<member name="remote_outbound_rtp" value="5" c:identifier="GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP" glib:nick="remote-outbound-rtp">
|
||||
<doc xml:space="preserve">remote-outbound-rtp</doc>
|
||||
</member>
|
||||
<member name="csrc" value="6" c:identifier="GST_WEBRTC_STATS_CSRC" glib:nick="csrc">
|
||||
<doc xml:space="preserve">csrc</doc>
|
||||
</member>
|
||||
<member name="peer_connection" value="7" c:identifier="GST_WEBRTC_STATS_PEER_CONNECTION" glib:nick="peer-connection">
|
||||
<doc xml:space="preserve">peer-connectiion</doc>
|
||||
</member>
|
||||
<member name="data_channel" value="8" c:identifier="GST_WEBRTC_STATS_DATA_CHANNEL" glib:nick="data-channel">
|
||||
<doc xml:space="preserve">data-channel</doc>
|
||||
</member>
|
||||
<member name="stream" value="9" c:identifier="GST_WEBRTC_STATS_STREAM" glib:nick="stream">
|
||||
<doc xml:space="preserve">stream</doc>
|
||||
</member>
|
||||
<member name="transport" value="10" c:identifier="GST_WEBRTC_STATS_TRANSPORT" glib:nick="transport">
|
||||
<doc xml:space="preserve">transport</doc>
|
||||
</member>
|
||||
<member name="candidate_pair" value="11" c:identifier="GST_WEBRTC_STATS_CANDIDATE_PAIR" glib:nick="candidate-pair">
|
||||
<doc xml:space="preserve">candidate-pair</doc>
|
||||
</member>
|
||||
<member name="local_candidate" value="12" c:identifier="GST_WEBRTC_STATS_LOCAL_CANDIDATE" glib:nick="local-candidate">
|
||||
<doc xml:space="preserve">local-candidate</doc>
|
||||
</member>
|
||||
<member name="remote_candidate" value="13" c:identifier="GST_WEBRTC_STATS_REMOTE_CANDIDATE" glib:nick="remote-candidate">
|
||||
<doc xml:space="preserve">remote-candidate</doc>
|
||||
</member>
|
||||
<member name="certificate" value="14" c:identifier="GST_WEBRTC_STATS_CERTIFICATE" glib:nick="certificate">
|
||||
<doc xml:space="preserve">certificate</doc>
|
||||
</member>
|
||||
</enumeration>
|
||||
<function name="webrtc_sdp_type_to_string" c:identifier="gst_webrtc_sdp_type_to_string" moved-to="WebRTCSDPType.to_string">
|
||||
|
|
Loading…
Reference in a new issue