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gst/audiorate/gstaudiorate.c: Don't rely on incoming buffers offset anymore, since it is completely broken when using...

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Original commit message from CVS:
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Don't rely on incoming buffers offset anymore, since it is completely
broken when using multiple segments.
Instead convert the incoming buffers timestamp to running time, and
then convert that value to the offsets.
Also inform GstSegment of the last outputted stop position, which is
needed if we received several segments with an unknown stop value.
This commit is contained in:
Edward Hervey 2006-08-29 10:32:34 +00:00
parent 1eb2a25683
commit 317bb22aca
2 changed files with 29 additions and 14 deletions
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10
ChangeLog
View file

@ -1,3 +1,13 @@
2006-08-29 Edward Hervey <edward@fluendo.com>
* gst/audiorate/gstaudiorate.c: (gst_audio_rate_chain):
Don't rely on incoming buffers offset anymore, since it is completely
broken when using multiple segments.
Instead convert the incoming buffers timestamp to running time, and
then convert that value to the offsets.
Also inform GstSegment of the last outputted stop position, which is
needed if we received several segments with an unknown stop value.
2006-08-29 Thomas Vander Stichele <thomas at apestaart dot org>
* ext/vorbis/vorbisenc.c: (gst_vorbis_enc_chain):

33
gst/audiorate/gstaudiorate.c
View file

@ -366,7 +366,7 @@ static GstFlowReturn
gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
{
GstAudioRate *audiorate;
GstClockTime in_time, in_duration;
GstClockTime in_time, in_duration, run_time;
guint64 in_offset, in_offset_end;
guint in_size;
GstFlowReturn ret = GST_FLOW_OK;
@ -404,22 +404,22 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
in_time = GST_BUFFER_TIMESTAMP (buf);
in_duration = GST_BUFFER_DURATION (buf);
in_size = GST_BUFFER_SIZE (buf);
in_offset = GST_BUFFER_OFFSET (buf);
in_offset_end = GST_BUFFER_OFFSET_END (buf);
/* don't really on buffer's offset */
/* We instead figure out using the runningtime version of the incoming buffer timestamp */
run_time =
gst_segment_to_running_time (&audiorate->segment, GST_FORMAT_TIME,
in_time);
in_offset = gst_util_uint64_scale_int (run_time, audiorate->rate, GST_SECOND);
in_offset_end = in_offset + in_size / audiorate->bytes_per_sample;
GST_LOG_OBJECT (audiorate,
"in_time:%" GST_TIME_FORMAT ", in_duration:%" GST_TIME_FORMAT
"in_time:%" GST_TIME_FORMAT ", run_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);
}
GST_TIME_ARGS (in_time), GST_TIME_ARGS (run_time),
GST_TIME_ARGS (in_duration), in_size, in_offset, in_offset_end,
audiorate->next_offset);
/* do we need to insert samples */
if (in_offset > audiorate->next_offset) {
@ -511,6 +511,11 @@ gst_audio_rate_chain (GstPad * pad, GstBuffer * buf)
buf = gst_buffer_make_metadata_writable (buf);
GST_BUFFER_FLAG_UNSET (buf, GST_BUFFER_FLAG_DISCONT);
}
/* set last_stop on segment */
gst_segment_set_last_stop (&audiorate->segment, GST_FORMAT_TIME,
GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf));
ret = gst_pad_push (audiorate->srcpad, buf);
audiorate->out++;