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cefsrc: Apply running time on audio buffers
Also add an audio meta to buffers, for good measure, and set discont flag when needed. This should help with A/V sync issues and audio cracks reported in #59.
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Original file line number | Diff line number | Diff line change |
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@@ -6,6 +6,7 @@ | |
#define CEF_VIDEO_CAPS "video/x-raw, format=BGRA, width=[1, 2147483647], height=[1, 2147483647], framerate=[1/1, 60/1], pixel-aspect-ratio=1/1" | ||
#define CEF_AUDIO_CAPS "audio/x-raw, format=F32LE, rate=[1, 2147483647], channels=[1, 2147483647], layout=interleaved" | ||
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#define gst_cef_demux_parent_class parent_class | ||
G_DEFINE_TYPE (GstCefDemux, gst_cef_demux, GST_TYPE_ELEMENT); | ||
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static GstStaticPadTemplate gst_cef_demux_sink_template = | ||
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@@ -58,6 +59,7 @@ gst_cef_demux_push_events (GstCefDemux *demux) | |
"channels", G_TYPE_INT, 2, | ||
"layout", G_TYPE_STRING, "interleaved", | ||
NULL); | ||
gst_audio_info_from_caps (&demux->audio_info, audio_caps); | ||
gst_pad_push_event (demux->asrcpad, gst_event_new_caps (audio_caps)); | ||
gst_caps_unref (audio_caps); | ||
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@@ -83,13 +85,55 @@ typedef struct | |
GstFlowReturn combined; | ||
} AudioPushData; | ||
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#if !GST_CHECK_VERSION(1, 18, 0) | ||
static GstClockTime | ||
gst_element_get_current_running_time (GstElement * element) | ||
{ | ||
GstClockTime base_time, clock_time; | ||
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g_return_val_if_fail (GST_IS_ELEMENT (element), GST_CLOCK_TIME_NONE); | ||
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base_time = gst_element_get_base_time (element); | ||
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if (!GST_CLOCK_TIME_IS_VALID (base_time)) { | ||
GST_DEBUG_OBJECT (element, "Could not determine base time"); | ||
return GST_CLOCK_TIME_NONE; | ||
} | ||
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clock_time = gst_element_get_current_clock_time (element); | ||
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if (!GST_CLOCK_TIME_IS_VALID (clock_time)) { | ||
return GST_CLOCK_TIME_NONE; | ||
} | ||
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if (clock_time < base_time) { | ||
GST_DEBUG_OBJECT (element, "Got negative current running time"); | ||
return GST_CLOCK_TIME_NONE; | ||
} | ||
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return clock_time - base_time; | ||
} | ||
#endif | ||
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static gboolean | ||
gst_cef_demux_push_audio_buffer (GstBuffer **buffer, guint idx, AudioPushData *push_data) | ||
{ | ||
GST_BUFFER_PTS (*buffer) += push_data->demux->ts_offset; | ||
push_data->demux->last_audio_time = gst_element_get_current_running_time (GST_ELEMENT_CAST (push_data->demux)); | ||
GST_BUFFER_DTS (*buffer) = push_data->demux->last_audio_time; | ||
GST_BUFFER_PTS (*buffer) = push_data->demux->last_audio_time; | ||
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gst_buffer_add_audio_meta (*buffer, &push_data->demux->audio_info, gst_buffer_get_size (*buffer), NULL); | ||
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GST_BUFFER_FLAG_UNSET (*buffer, GST_BUFFER_FLAG_DISCONT); | ||
if (push_data->demux->need_discont) { | ||
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT); | ||
push_data->demux->need_discont = FALSE; | ||
} | ||
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push_data->combined = gst_flow_combiner_update_pad_flow (push_data->flow_combiner, push_data->demux->asrcpad, | ||
gst_pad_push (push_data->demux->asrcpad, *buffer)); | ||
push_data->demux->last_audio_time = GST_BUFFER_PTS (*buffer) + GST_BUFFER_DURATION (*buffer); | ||
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*buffer = NULL; | ||
return TRUE; | ||
} | ||
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@@ -127,11 +171,6 @@ gst_cef_demux_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) | |
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gst_cef_demux_push_events (demux); | ||
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if (!GST_CLOCK_TIME_IS_VALID (demux->ts_offset)) { | ||
demux->ts_offset = GST_BUFFER_PTS (buffer); | ||
} | ||
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for (tmp = demux->cef_audio_stream_start_events; tmp; tmp = tmp->next) { | ||
const GstStructure *s = gst_event_get_structure ((GstEvent *) tmp->data); | ||
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@@ -228,6 +267,31 @@ gst_cef_demux_sink_query (GstPad *pad, GstObject *parent, GstQuery *query) | |
return ret; | ||
} | ||
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static GstStateChangeReturn | ||
gst_cef_demux_change_state (GstElement * element, GstStateChange transition) | ||
{ | ||
GstStateChangeReturn result; | ||
GstCefDemux *demux = (GstCefDemux *) element; | ||
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GST_DEBUG_OBJECT (demux, "%s", gst_state_change_get_name (transition)); | ||
result = GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS , change_state, (element, transition), GST_STATE_CHANGE_FAILURE); | ||
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switch (transition) { | ||
case GST_STATE_CHANGE_PAUSED_TO_READY: | ||
gst_flow_combiner_reset (demux->flow_combiner); | ||
break; | ||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED: | ||
demux->need_discont = TRUE; | ||
break; | ||
default: | ||
break; | ||
} | ||
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return result; | ||
} | ||
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static void | ||
gst_cef_demux_init (GstCefDemux * demux) | ||
{ | ||
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@@ -248,11 +312,13 @@ gst_cef_demux_init (GstCefDemux * demux) | |
gst_element_add_pad (GST_ELEMENT (demux), demux->asrcpad); | ||
gst_flow_combiner_add_pad (demux->flow_combiner, demux->asrcpad); | ||
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gst_audio_info_init (&demux->audio_info); | ||
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demux->need_stream_start = TRUE; | ||
demux->need_caps = TRUE; | ||
demux->need_segment = TRUE; | ||
demux->last_audio_time = 0; | ||
demux->ts_offset = GST_CLOCK_TIME_NONE; | ||
demux->need_discont = TRUE; | ||
demux->last_audio_time = GST_CLOCK_TIME_NONE; | ||
} | ||
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static void | ||
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@@ -273,6 +339,8 @@ gst_cef_demux_class_init (GstCefDemuxClass * klass) | |
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gobject_class->finalize = gst_cef_demux_finalize; | ||
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gstelement_class->change_state = GST_DEBUG_FUNCPTR(gst_cef_demux_change_state); | ||
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gst_element_class_set_static_metadata (gstelement_class, | ||
"Chromium Embedded Framework demuxer", "Demuxer/Audio/Video", | ||
"Demuxes audio and video from cefsrc", "Mathieu Duponchelle <[email protected]>"); | ||
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