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gst/audiorate/gstaudiorate.c: Fix audiorate some more.
Original commit message from CVS: * gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset), (gst_audio_rate_setcaps), (gst_audio_rate_init), (gst_audio_rate_sink_event), (gst_audio_rate_src_event), (gst_audio_rate_chain), (gst_audio_rate_change_state): Fix audiorate some more. Reset and resync counters on flush and READY. Handle the DISCONT flag correctly. Use GstSegment to track position. Fail when not negotiated.
This commit is contained in:
Wim Taymans
parent
fe2c1e2d02
commit
0fc6e3d087
12
ChangeLog
12
ChangeLog
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@ -1,3 +1,15 @@
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2006-08-28 Wim Taymans <wim@fluendo.com>
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* gst/audiorate/gstaudiorate.c: (gst_audio_rate_reset),
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(gst_audio_rate_setcaps), (gst_audio_rate_init),
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(gst_audio_rate_sink_event), (gst_audio_rate_src_event),
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(gst_audio_rate_chain), (gst_audio_rate_change_state):
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Fix audiorate some more.
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Reset and resync counters on flush and READY.
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Handle the DISCONT flag correctly.
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Use GstSegment to track position.
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Fail when not negotiated.
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2006-08-25 Michael Smith <msmith@fluendo.com>
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* gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_render):
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@ -48,14 +48,20 @@ struct _GstAudioRate
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GstPad *sinkpad, *srcpad;
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/* audio format */
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gint bytes_per_sample;
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gint rate;
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/* stats */
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guint64 in, out, add, drop;
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gboolean silent;
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/* audio state */
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guint64 offset;
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guint64 next_offset;
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guint64 in, out, add, drop;
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gboolean silent;
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gboolean discont;
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GstSegment segment;
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};
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struct _GstAudioRateClass
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@ -109,6 +115,8 @@ static GstStaticPadTemplate gst_audio_rate_sink_template =
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static void gst_audio_rate_base_init (gpointer g_class);
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static void gst_audio_rate_class_init (GstAudioRateClass * klass);
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static void gst_audio_rate_init (GstAudioRate * audiorate);
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static gboolean gst_audio_rate_sink_event (GstPad * pad, GstEvent * event);
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static gboolean gst_audio_rate_src_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_audio_rate_chain (GstPad * pad, GstBuffer * buf);
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static void gst_audio_rate_set_property (GObject * object,
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@ -191,6 +199,17 @@ gst_audio_rate_class_init (GstAudioRateClass * klass)
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element_class->change_state = gst_audio_rate_change_state;
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}
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static void
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gst_audio_rate_reset (GstAudioRate * audiorate)
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{
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audiorate->offset = -1;
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audiorate->next_offset = -1;
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audiorate->discont = TRUE;
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gst_segment_init (&audiorate->segment, GST_FORMAT_UNDEFINED);
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GST_DEBUG_OBJECT (audiorate, "handle reset");
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}
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static gboolean
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gst_audio_rate_setcaps (GstPad * pad, GstCaps * caps)
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{
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@ -198,31 +217,46 @@ gst_audio_rate_setcaps (GstPad * pad, GstCaps * caps)
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GstStructure *structure;
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GstPad *otherpad;
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gboolean ret = FALSE;
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gint channels, width;
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gint channels, width, rate;
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audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
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otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
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audiorate->srcpad;
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if (!gst_pad_set_caps (otherpad, caps))
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goto beach;
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structure = gst_caps_get_structure (caps, 0);
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if (!gst_structure_get_int (structure, "channels", &channels) ||
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!gst_structure_get_int (structure, "width", &width)) {
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goto beach;
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}
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if (!gst_structure_get_int (structure, "channels", &channels))
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goto wrong_caps;
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if (!gst_structure_get_int (structure, "width", &width))
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goto wrong_caps;
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if (!gst_structure_get_int (structure, "rate", &rate))
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goto wrong_caps;
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audiorate->bytes_per_sample = channels * (width / 8);
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if (audiorate->bytes_per_sample == 0)
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audiorate->bytes_per_sample = 1;
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ret = TRUE;
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goto wrong_format;
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beach:
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audiorate->rate = rate;
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/* the format is correct, configure caps on other pad */
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otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad :
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audiorate->srcpad;
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ret = gst_pad_set_caps (otherpad, caps);
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done:
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gst_object_unref (audiorate);
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return ret;
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/* ERRORS */
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wrong_caps:
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{
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GST_DEBUG_OBJECT (audiorate, "could not get channels/width from caps");
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goto done;
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}
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wrong_format:
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{
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GST_DEBUG_OBJECT (audiorate, "bytes_per_samples gave 0");
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goto done;
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}
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}
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static void
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@ -230,18 +264,19 @@ gst_audio_rate_init (GstAudioRate * audiorate)
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{
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audiorate->sinkpad =
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gst_pad_new_from_static_template (&gst_audio_rate_sink_template, "sink");
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gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
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gst_pad_set_event_function (audiorate->sinkpad, gst_audio_rate_sink_event);
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gst_pad_set_chain_function (audiorate->sinkpad, gst_audio_rate_chain);
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gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audio_rate_setcaps);
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gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps);
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gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad);
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audiorate->srcpad =
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gst_pad_new_from_static_template (&gst_audio_rate_src_template, "src");
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gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
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gst_pad_set_event_function (audiorate->srcpad, gst_audio_rate_src_event);
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gst_pad_set_setcaps_function (audiorate->srcpad, gst_audio_rate_setcaps);
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gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps);
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gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad);
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audiorate->bytes_per_sample = 1;
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audiorate->in = 0;
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audiorate->out = 0;
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audiorate->drop = 0;
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@ -249,6 +284,84 @@ gst_audio_rate_init (GstAudioRate * audiorate)
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audiorate->silent = DEFAULT_SILENT;
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}
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static gboolean
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gst_audio_rate_sink_event (GstPad * pad, GstEvent * event)
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{
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gboolean res;
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GstAudioRate *audiorate;
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audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_STOP:
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GST_DEBUG_OBJECT (audiorate, "handling FLUSH_STOP");
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gst_audio_rate_reset (audiorate);
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res = gst_pad_push_event (audiorate->srcpad, event);
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break;
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case GST_EVENT_NEWSEGMENT:
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{
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GstFormat format;
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gdouble rate, arate;
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gint64 start, stop, time;
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gboolean update;
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gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
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&start, &stop, &time);
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GST_DEBUG_OBJECT (audiorate, "handle NEWSEGMENT");
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/* FIXME:
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* - sparse stream support. For this, the update flag is TRUE and the
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* start/time positions are updated, meaning that time progressed by
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* time - old_time amount and we need to fill that gap with empty
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* samples.
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* - fill the current segment if it has a valid stop position. This
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* happens when the update flag is FALSE. With the segment helper we can
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* calculate the accumulated time and compare this to the next_offset.
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*/
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if (!update) {
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/* a new segment starts. We need to figure out what will be the next
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* sample offset. We mark the offsets as invalid so that the _chain
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* function will perform this calculation. */
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audiorate->offset = -1;
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audiorate->next_offset = -1;
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}
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gst_segment_set_newsegment_full (&audiorate->segment, update, rate, arate,
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format, start, stop, time);
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res = gst_pad_push_event (audiorate->srcpad, event);
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break;
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}
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case GST_EVENT_EOS:
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default:
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res = gst_pad_push_event (audiorate->srcpad, event);
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break;
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}
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gst_object_unref (audiorate);
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return res;
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}
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static gboolean
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gst_audio_rate_src_event (GstPad * pad, GstEvent * event)
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{
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gboolean res;
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GstAudioRate *audiorate;
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audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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default:
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res = gst_pad_push_event (audiorate->sinkpad, event);
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break;
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}
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gst_object_unref (audiorate);
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return res;
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}
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static GstFlowReturn
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gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
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{
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@ -260,6 +373,32 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
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audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
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if (audiorate->bytes_per_sample == 0)
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goto not_negotiated;
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if (audiorate->offset == -1) {
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gint64 pos;
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/* first buffer, we are negotiated and we have a segment, calculate the
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* current expected offsets based on the segment.time, which is the first
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* media time of the segment and should match the media time of the first
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* buffer in that segment, which is the offset expressed in DEFAULT units.
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*/
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pos = audiorate->segment.time;
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if (pos != 0) {
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if (audiorate->segment.format == GST_FORMAT_TIME) {
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/* convert first timestamp of segment to sample position */
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pos = gst_util_uint64_scale_int (pos, audiorate->rate, GST_SECOND);
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} else {
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/* FIXME, we don't know, start from 0 then... */
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pos = 0;
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}
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}
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GST_DEBUG_OBJECT (audiorate, "resync to offset %" G_GINT64_FORMAT, pos);
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audiorate->offset = pos;
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audiorate->next_offset = pos;
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}
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audiorate->in++;
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in_time = GST_BUFFER_TIMESTAMP (buf);
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@ -301,7 +440,17 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
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GST_BUFFER_OFFSET (fill) = audiorate->next_offset;
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GST_BUFFER_OFFSET_END (fill) = in_offset;
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if ((ret = gst_pad_push (audiorate->srcpad, fill) != GST_FLOW_OK))
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/* we created this buffer to filla gap */
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GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_GAP);
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/* set discont if it's pending, this is mostly done for the first buffer and
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* after a flushing seek */
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if (audiorate->discont) {
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GST_BUFFER_FLAG_SET (fill, GST_BUFFER_FLAG_DISCONT);
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audiorate->discont = FALSE;
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}
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ret = gst_pad_push (audiorate->srcpad, fill);
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if (ret != GST_FLOW_OK)
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goto beach;
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audiorate->out++;
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audiorate->add += fillsamples;
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@ -349,13 +498,36 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
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audiorate->drop += truncsamples;
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}
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}
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if (audiorate->discont) {
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/* we need to output a discont buffer, do so now */
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GST_DEBUG_OBJECT (audiorate, "marking DISCONT on output buffer");
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GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
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audiorate->discont = FALSE;
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} else if (GST_BUFFER_IS_DISCONT (buf)) {
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/* else we make everything continuous so we can safely remove the DISCONT
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* flag from the buffer if there was one */
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GST_DEBUG_OBJECT (audiorate, "removing DISCONT from buffer");
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buf = gst_buffer_make_metadata_writable (buf);
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GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
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}
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ret = gst_pad_push (audiorate->srcpad, buf);
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audiorate->out++;
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audiorate->next_offset = in_offset_end;
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beach:
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audiorate->offset += in_size / audiorate->bytes_per_sample;
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gst_object_unref (audiorate);
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return ret;
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/* ERRORS */
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not_negotiated:
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{
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GST_ELEMENT_ERROR (audiorate, STREAM, FORMAT,
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(NULL), ("pipeline error, format was not negotiated"));
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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static void
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@ -411,8 +583,12 @@ gst_audio_rate_change_state (GstElement * element, GstStateChange transition)
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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audiorate->offset = 0;
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audiorate->next_offset = 0;
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audiorate->in = 0;
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audiorate->out = 0;
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audiorate->drop = 0;
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audiorate->bytes_per_sample = 0;
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audiorate->add = 0;
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gst_audio_rate_reset (audiorate);
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break;
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default:
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break;
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