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audiobuffersplit: Combine two if expressions to reduce indentation

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2780>
This commit is contained in:
Sebastian Dröge 2022-07-21 13:38:22 +03:00 committed by GStreamer Marge Bot
parent 0cadd10474
commit 0485c354d2
1 changed files with 62 additions and 66 deletions
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128
subprojects/gst-plugins-bad/gst/audiobuffersplit/gstaudiobuffersplit.c
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@ -540,84 +540,80 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
avail_samples, GST_SECOND, rate * self->in_segment.rate);
}
if (self->gapless) {
if (self->current_offset != -1) {
GST_DEBUG_OBJECT (self,
"Got discont in gapless mode: Current running time %" GST_TIME_FORMAT
", current end running time %" GST_TIME_FORMAT
", running time after discont %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_rt),
GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt));
if (self->gapless && self->current_offset != -1) {
GST_DEBUG_OBJECT (self,
"Got discont in gapless mode: Current running time %" GST_TIME_FORMAT
", current end running time %" GST_TIME_FORMAT
", running time after discont %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_rt),
GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt));
new_offset =
gst_util_uint64_scale (current_rt - self->resync_rt,
rate * ABS (self->in_segment.rate), GST_SECOND);
if (current_rt < self->resync_rt) {
guint64 drop_samples;
new_offset =
gst_util_uint64_scale (current_rt - self->resync_rt,
rate * ABS (self->in_segment.rate), GST_SECOND);
if (current_rt < self->resync_rt) {
guint64 drop_samples;
gst_util_uint64_scale (self->resync_rt -
current_rt, rate * ABS (self->in_segment.rate), GST_SECOND);
drop_samples = self->current_offset + avail_samples + new_offset;
new_offset =
gst_util_uint64_scale (self->resync_rt -
current_rt, rate * ABS (self->in_segment.rate), GST_SECOND);
drop_samples = self->current_offset + avail_samples + new_offset;
GST_DEBUG_OBJECT (self,
"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
GST_SECOND, rate)));
discont = FALSE;
} else if (new_offset > self->current_offset + avail_samples) {
guint64 silence_samples =
new_offset - (self->current_offset + avail_samples);
const GstAudioFormatInfo *info = gst_audio_format_get_info (format);
GstClockTime silence_time =
gst_util_uint64_scale (silence_samples, GST_SECOND, rate);
if (silence_time > self->max_silence_time) {
GST_DEBUG_OBJECT (self,
"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
GST_SECOND, rate)));
discont = FALSE;
} else if (new_offset > self->current_offset + avail_samples) {
guint64 silence_samples =
new_offset - (self->current_offset + avail_samples);
const GstAudioFormatInfo *info = gst_audio_format_get_info (format);
GstClockTime silence_time =
gst_util_uint64_scale (silence_samples, GST_SECOND, rate);
"Not inserting %" G_GUINT64_FORMAT " samples of silence (%"
GST_TIME_FORMAT " exceeds maximum %" GST_TIME_FORMAT ")",
silence_samples, GST_TIME_ARGS (silence_time),
GST_TIME_ARGS (self->max_silence_time));
} else {
GST_DEBUG_OBJECT (self,
"Inserting %" G_GUINT64_FORMAT " samples of silence (%"
GST_TIME_FORMAT ")", silence_samples, GST_TIME_ARGS (silence_time));
if (silence_time > self->max_silence_time) {
GST_DEBUG_OBJECT (self,
"Not inserting %" G_GUINT64_FORMAT " samples of silence (%"
GST_TIME_FORMAT " exceeds maximum %" GST_TIME_FORMAT ")",
silence_samples, GST_TIME_ARGS (silence_time),
GST_TIME_ARGS (self->max_silence_time));
} else {
GST_DEBUG_OBJECT (self,
"Inserting %" G_GUINT64_FORMAT " samples of silence (%"
GST_TIME_FORMAT ")", silence_samples,
GST_TIME_ARGS (silence_time));
/* Insert silence buffers to fill the gap in 1s chunks */
while (silence_samples > 0) {
guint n_samples = MIN (silence_samples, rate);
GstBuffer *silence;
GstMapInfo map;
/* Insert silence buffers to fill the gap in 1s chunks */
while (silence_samples > 0) {
guint n_samples = MIN (silence_samples, rate);
GstBuffer *silence;
GstMapInfo map;
silence = gst_buffer_new_and_alloc (n_samples * bpf);
GST_BUFFER_FLAG_SET (silence, GST_BUFFER_FLAG_GAP);
gst_buffer_map (silence, &map, GST_MAP_WRITE);
gst_audio_format_info_fill_silence (info, map.data, map.size);
gst_buffer_unmap (silence, &map);
silence = gst_buffer_new_and_alloc (n_samples * bpf);
GST_BUFFER_FLAG_SET (silence, GST_BUFFER_FLAG_GAP);
gst_buffer_map (silence, &map, GST_MAP_WRITE);
gst_audio_format_info_fill_silence (info, map.data, map.size);
gst_buffer_unmap (silence, &map);
gst_adapter_push (self->adapter, silence);
ret =
gst_audio_buffer_split_output (self, FALSE, rate, bpf,
samples_per_buffer);
if (ret != GST_FLOW_OK)
return ret;
gst_adapter_push (self->adapter, silence);
ret =
gst_audio_buffer_split_output (self, FALSE, rate, bpf,
samples_per_buffer);
if (ret != GST_FLOW_OK)
return ret;
silence_samples -= n_samples;
}
discont = FALSE;
silence_samples -= n_samples;
}
} else if (new_offset < self->current_offset + avail_samples) {
guint64 drop_samples =
self->current_offset + avail_samples - new_offset;
GST_DEBUG_OBJECT (self,
"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
GST_SECOND, rate)));
self->drop_samples = drop_samples;
discont = FALSE;
}
} else if (new_offset < self->current_offset + avail_samples) {
guint64 drop_samples = self->current_offset + avail_samples - new_offset;
GST_DEBUG_OBJECT (self,
"Dropping %" G_GUINT64_FORMAT " samples (%" GST_TIME_FORMAT ")",
drop_samples, GST_TIME_ARGS (gst_util_uint64_scale (drop_samples,
GST_SECOND, rate)));
self->drop_samples = drop_samples;
discont = FALSE;
}
}