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gst/audiorate/gstaudiorate.c: Add debugging category.

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Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
(gst_audio_rate_change_state), (plugin_init):
Add debugging category.
Fix type issues.
Add case for incoming buffers without valid offset/offset_end.
This commit is contained in:
Edward Hervey 2006-01-10 15:47:48 +00:00
parent b0c21cab17
commit 927b499e9d
2 changed files with 31 additions and 3 deletions
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8
ChangeLog
View file

@ -1,3 +1,11 @@
2006-01-10 Edward Hervey <edward@fluendo.com>
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain),
(gst_audio_rate_change_state), (plugin_init):
Add debugging category.
Fix type issues.
Add case for incoming buffers without valid offset/offset_end.
2006-01-10 Michael Smith <msmith@fluendo.com>
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_dispose):

26
gst/audiorate/gstaudiorate.c
View file

@ -25,6 +25,9 @@
#include <gst/gst.h>
#include <gst/audio/audio.h>
GST_DEBUG_CATEGORY (audio_rate_debug);
#define GST_CAT_DEFAULT audio_rate_debug
#define GST_TYPE_AUDIO_RATE \
(gst_audio_rate_get_type())
#define GST_AUDIO_RATE(obj) \
@ -48,6 +51,7 @@ struct _GstAudioRate
gint bytes_per_sample;
/* audio state */
guint64 offset;
guint64 next_offset;
guint64 in, out, add, drop;
@ -248,7 +252,7 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
GstAudioRate *audiorate;
GstClockTime in_time, in_duration;
guint64 in_offset, in_offset_end;
gint in_size;
guint in_size;
GstFlowReturn ret = GST_FLOW_OK;
audiorate = GST_AUDIO_RATE (gst_pad_get_parent (pad));
@ -261,8 +265,18 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
in_offset = GST_BUFFER_OFFSET (buf);
in_offset_end = GST_BUFFER_OFFSET_END (buf);
GST_LOG_OBJECT (audiorate,
"in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT
", in_size:%u, in_offset:%lld, in_offset_end:%lld" ", ->next_offset:%lld",
GST_TIME_ARGS (in_time), GST_TIME_ARGS (in_duration), in_size, in_offset,
in_offset_end, audiorate->next_offset);
if (in_offset == GST_CLOCK_TIME_NONE || in_offset_end == GST_CLOCK_TIME_NONE) {
GST_WARNING_OBJECT (audiorate, "audiorate got buffer without offsets");
in_offset = audiorate->offset;
in_offset_end = audiorate->offset + in_size / audiorate->bytes_per_sample;
GST_WARNING_OBJECT (audiorate, "in_offset:%lld, in_offset_end:%lld",
in_offset, in_offset_end);
}
/* do we need to insert samples */
@ -308,7 +322,8 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
goto beach;
} else {
guint64 truncsamples, truncsize, leftsize;
guint64 truncsamples;
guint truncsize, leftsize;
GstBuffer *trunc;
/* truncate buffer */
@ -316,7 +331,7 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
truncsize = truncsamples * audiorate->bytes_per_sample;
leftsize = in_size - truncsize;
trunc = gst_buffer_create_sub (buf, truncsize, in_size);
trunc = gst_buffer_create_sub (buf, truncsize, leftsize);
GST_BUFFER_DURATION (trunc) = in_duration * leftsize / in_size;
GST_BUFFER_TIMESTAMP (trunc) =
in_time + in_duration - GST_BUFFER_DURATION (trunc);
@ -336,6 +351,7 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
audiorate->next_offset = in_offset_end;
beach:
audiorate->offset += in_size / audiorate->bytes_per_sample;
return ret;
}
@ -392,6 +408,7 @@ gst_audio_rate_change_state (GstElement * element, GstStateChange transition)
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
audiorate->offset = 0;
audiorate->next_offset = 0;
break;
default:
@ -407,6 +424,9 @@ gst_audio_rate_change_state (GstElement * element, GstStateChange transition)
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (audio_rate_debug, "audiorate", 0,
"AudioRate stream fixer");
return gst_element_register (plugin, "audiorate", GST_RANK_NONE,
GST_TYPE_AUDIO_RATE);
}