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libs: audio: implement support for non-interleaved audio in gst_audio_buffer_clip()

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=796740
This commit is contained in:
George Kiagiadakis 2018-02-08 14:34:23 +02:00
parent 5518eb26fb
commit 9cb09e7269
1 changed files with 40 additions and 10 deletions
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50
gst-libs/gst/audio/audio.c
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@ -80,6 +80,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
gint rate, gint bpf)
{
GstBuffer *ret;
GstAudioMeta *meta;
GstClockTime timestamp = GST_CLOCK_TIME_NONE, duration = GST_CLOCK_TIME_NONE;
guint64 offset = GST_BUFFER_OFFSET_NONE, offset_end = GST_BUFFER_OFFSET_NONE;
gsize trim, size, osize;
@ -98,8 +99,11 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
* Calculate the missing values for the calculations,
* they won't be changed later though. */
meta = gst_buffer_get_audio_meta (buffer);
/* these variables measure samples */
trim = 0;
osize = size = gst_buffer_get_size (buffer);
osize = size = meta ? meta->samples : (gst_buffer_get_size (buffer) / bpf);
/* no data, nothing to clip */
if (!size)
@ -111,7 +115,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
duration = GST_BUFFER_DURATION (buffer);
} else {
change_duration = FALSE;
duration = gst_util_uint64_scale (size / bpf, GST_SECOND, rate);
duration = gst_util_uint64_scale (size, GST_SECOND, rate);
}
if (GST_BUFFER_OFFSET_IS_VALID (buffer)) {
@ -125,7 +129,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
offset_end = GST_BUFFER_OFFSET_END (buffer);
} else {
change_offset_end = FALSE;
offset_end = offset + size / bpf;
offset_end = offset + size;
}
if (segment->format == GST_FORMAT_TIME) {
@ -149,8 +153,8 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset)
offset += diff;
trim += diff * bpf;
size -= diff * bpf;
trim += diff;
size -= diff;
}
diff = stop - cstop;
@ -161,7 +165,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
diff = gst_util_uint64_scale (diff, rate, GST_SECOND);
if (change_offset_end)
offset_end -= diff;
size -= diff * bpf;
size -= diff;
}
} else {
gst_buffer_unref (buffer);
@ -188,8 +192,8 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
trim += diff * bpf;
size -= diff * bpf;
trim += diff;
size -= diff;
}
diff = stop - cstop;
@ -199,7 +203,7 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
if (change_duration)
duration -= gst_util_uint64_scale (diff, GST_SECOND, rate);
size -= diff * bpf;
size -= diff;
}
} else {
gst_buffer_unref (buffer);
@ -218,8 +222,34 @@ gst_audio_buffer_clip (GstBuffer * buffer, const GstSegment * segment,
ret = gst_buffer_make_writable (ret);
GST_BUFFER_DURATION (ret) = duration;
}
} else if (meta && meta->info.layout == GST_AUDIO_LAYOUT_NON_INTERLEAVED) {
/* modify only the meta to avoid making copies of the planes */
gint i;
ret = gst_buffer_make_writable (buffer);
meta = gst_buffer_get_audio_meta (buffer);
meta->samples = size;
for (i = 0; i < meta->info.channels; i++) {
meta->offsets[i] += trim * bpf / meta->info.channels;
}
GST_BUFFER_TIMESTAMP (ret) = timestamp;
if (change_duration)
GST_BUFFER_DURATION (ret) = duration;
if (change_offset)
GST_BUFFER_OFFSET (ret) = offset;
if (change_offset_end)
GST_BUFFER_OFFSET_END (ret) = offset_end;
} else {
/* Get a writable buffer and apply all changes */
/* resize the buffer, effectively cutting out all
* the samples that are no longer relevant */
/* convert samples to bytes */
trim *= bpf;
size *= bpf;
GST_DEBUG ("trim %" G_GSIZE_FORMAT " size %" G_GSIZE_FORMAT, trim, size);
ret = gst_buffer_copy_region (buffer, GST_BUFFER_COPY_ALL, trim, size);
gst_buffer_unref (buffer);