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audiorate: fix support for non-interleaved buffers

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=796741
This commit is contained in:
George Kiagiadakis 2018-02-12 14:45:44 +02:00
parent b33d70e97f
commit 09171b11ac
1 changed files with 12 additions and 14 deletions
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26
gst/audiorate/gstaudiorate.c
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@ -439,6 +439,7 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
GstFlowReturn ret = GST_FLOW_OK;
GstClockTimeDiff diff;
gint rate, bpf;
GstAudioMeta *meta;
audiorate = GST_AUDIO_RATE (parent);
@ -492,8 +493,9 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
in_time = audiorate->next_ts;
}
meta = gst_buffer_get_audio_meta (buf);
in_size = gst_buffer_get_size (buf);
in_samples = in_size / bpf;
in_samples = meta ? meta->samples : in_size / bpf;
audiorate->in += in_samples;
/* calculate the buffer offset */
@ -549,6 +551,10 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
fillmap.size);
gst_buffer_unmap (fill, &fillmap);
if (audiorate->info.layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
gst_buffer_add_audio_meta (fill, &audiorate->info, cursamples, NULL);
}
GST_DEBUG_OBJECT (audiorate, "inserting %" G_GUINT64_FORMAT " samples",
cursamples);
@ -591,7 +597,7 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
} else if (in_offset < audiorate->next_offset) {
/* need to remove samples */
if (in_offset_end <= audiorate->next_offset) {
guint64 drop = in_size / bpf;
guint64 drop = in_samples;
audiorate->drop += drop;
@ -607,24 +613,16 @@ gst_audio_rate_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
goto beach;
} else {
guint64 truncsamples;
guint truncsize, leftsize;
GstBuffer *trunc;
guint64 truncsamples, leftsamples;
/* truncate buffer */
truncsamples = audiorate->next_offset - in_offset;
truncsize = truncsamples * bpf;
leftsize = in_size - truncsize;
leftsamples = in_samples - truncsamples;
trunc =
gst_buffer_copy_region (buf, GST_BUFFER_COPY_ALL, truncsize,
leftsize);
gst_buffer_unref (buf);
buf = trunc;
buf = gst_audio_buffer_truncate (buf, bpf, truncsamples, leftsamples);
audiorate->drop += truncsamples;
audiorate->out += (leftsize / bpf);
audiorate->out += leftsamples;
GST_DEBUG_OBJECT (audiorate, "truncating %" G_GUINT64_FORMAT " samples",
truncsamples);