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interaudiosink: Ensure adapters don't store buffers with audio meta

Browse source 
The interaudiosrc might take buffers of different sizes from the audio adapter,
so keeping metas consistency would be an issue. So the sink now strips the audio
metas away and the src adds them back (for non-interleaved layouts only) when
taking buffers from the adapter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5324>
This commit is contained in:
Philippe Normand 2023-09-27 12:33:39 +02:00 committed by GStreamer Marge Bot
parent 46dbe2a372
commit 886bd7e4e0
3 changed files with 20 additions and 4 deletions
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1
subprojects/gst-integration-testsuites/testsuites/validate.testslist
View file

@ -936,6 +936,7 @@ validate.test.h264.parse.trickmode_predicted.seek_trickmode_predicted
validate.test.h264.x264enc_youtube_bitrate.fullhd_lowframerate
validate.test.h265parse.alternate
validate.test.h265parse.trickmode_predicted.seek_trickmode_predicted
validate.test.inter.audio-planar
validate.test.interlace.interlace_deinterlace
validate.test.interlace.interlace_deinterlace_alternate
validate.test.matroska.demux_flush_within_cluster.default

4
subprojects/gst-integration-testsuites/testsuites/validate/inter/audio-planar.validatetest Normal file
View file

@ -0,0 +1,4 @@
meta,
args = {
"audiotestsrc num-buffers=10 ! audio/x-raw,format=S16LE,channels=2,layout=non-interleaved ! interaudiosink interaudiosrc num-buffers=10 ! audioconvert ! audio/x-raw,layout=interleaved,format=S16LE,channels=2 ! queue ! fakeaudiosink",
}

19
subprojects/gst-plugins-bad/gst/inter/gstinteraudiosink.c
View file

@ -293,6 +293,7 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
guint n, bpf;
guint64 period_time, buffer_time;
guint64 period_samples, buffer_samples;
GstBuffer *tmp;
GST_DEBUG_OBJECT (interaudiosink, "render %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
@ -330,16 +331,26 @@ gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
n = gst_adapter_available (interaudiosink->input_adapter);
if (period_samples * bpf > gst_buffer_get_size (buffer) + n) {
gst_adapter_push (interaudiosink->input_adapter, gst_buffer_ref (buffer));
GstAudioMeta *audio_meta = NULL;
tmp = gst_buffer_copy_deep (buffer);
audio_meta = gst_buffer_get_audio_meta (tmp);
if (audio_meta != NULL)
gst_buffer_remove_meta (tmp, GST_META_CAST (audio_meta));
gst_adapter_push (interaudiosink->input_adapter, tmp);
} else {
GstBuffer *tmp;
GstAudioMeta *audio_meta = NULL;
if (n > 0) {
tmp = gst_adapter_take_buffer (interaudiosink->input_adapter, n);
gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
}
gst_adapter_push (interaudiosink->surface->audio_adapter,
gst_buffer_ref (buffer));
tmp = gst_buffer_copy_deep (buffer);
audio_meta = gst_buffer_get_audio_meta (tmp);
if (audio_meta != NULL)
gst_buffer_remove_meta (tmp, GST_META_CAST (audio_meta));
gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
}
g_mutex_unlock (&interaudiosink->surface->mutex);