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| author | Thomas Vander Stichele <thomas@apestaart.org> | 2005-08-25 17:20:02 +0000 |
|---|---|---|
| committer | Thomas Vander Stichele <thomas@apestaart.org> | 2005-08-25 17:20:02 +0000 |
| commit | 7d1ce173830af4c35c654569a232728139dd517e (patch) | |
| tree | ced3a8d447a190a5fbc71169da4613f20b27b5c2 /gst/audioresample/gstaudioresample.c | |
| parent | 1dfd2d885647ba0e5b8816f61a6a3617e5aeb834 (diff) | |
| download | gst-plugins-bad-7d1ce173830af4c35c654569a232728139dd517e.tar.gz gst-plugins-bad-7d1ce173830af4c35c654569a232728139dd517e.tar.bz2 gst-plugins-bad-7d1ce173830af4c35c654569a232728139dd517e.zip | |
check/: add a test for audioconvert
Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
Diffstat (limited to 'gst/audioresample/gstaudioresample.c')
| -rw-r--r-- | gst/audioresample/gstaudioresample.c | 12 |
1 files changed, 9 insertions, 3 deletions
diff --git a/gst/audioresample/gstaudioresample.c b/gst/audioresample/gstaudioresample.c index cbbfe81a..02537075 100644 --- a/gst/audioresample/gstaudioresample.c +++ b/gst/audioresample/gstaudioresample.c @@ -395,14 +395,20 @@ static GstFlowReturn GST_LOG_OBJECT (audioresample, "resample gave me %d bytes or %d samples", outsize, outsamples); + GST_BUFFER_OFFSET (outbuf) = audioresample->offset; GST_BUFFER_TIMESTAMP (outbuf) = audioresample->offset * GST_SECOND / audioresample->o_rate; - GST_BUFFER_DURATION (outbuf) = - outsamples * GST_SECOND / audioresample->o_rate; - GST_BUFFER_OFFSET (outbuf) = audioresample->offset; + audioresample->offset += outsamples; GST_BUFFER_OFFSET_END (outbuf) = audioresample->offset; + /* we calculate DURATION as the difference between "next" timestamp + * and current timestamp so we ensure a contiguous stream, instead of + * having rounding errors. */ + GST_BUFFER_DURATION (outbuf) = + audioresample->offset * GST_SECOND / audioresample->o_rate - + GST_BUFFER_TIMESTAMP (outbuf); + /* check for possible mem corruption */ if (outsize > GST_BUFFER_SIZE (outbuf)) { /* this is an error that when it happens, would need fixing in the |