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audioconvert: handle new GstRequestMixMatrix custom upstream event

Browse source 
An example use case is the gstwebrtc-api demo, which will cause
webrtcsink to forward such events. This lets the end user define a mix
matrix without requiring any application code server side.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7363>
This commit is contained in:
Mathieu Duponchelle 2024-08-15 17:01:31 +02:00 committed by GStreamer Marge Bot
parent fd90d7bdee
commit 25f3ab2e6c
1 changed files with 48 additions and 4 deletions
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52
subprojects/gst-plugins-base/gst/audioconvert/gstaudioconvert.c
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@ -71,6 +71,16 @@
* g_value_unset (&v);
* ]|
*
* The mix matrix can also be passed through a custom upstream event:
*
* |[
* GstStructure *s = gst_structure_new("GstRequestAudioMixMatrix", "matrix", GST_TYPE_ARRAY, &v, NULL);
* GstEvent *event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, s);
* GstPad *srcpad = gst_element_get_static_pad(audioconvert, "src");
* gst_pad_send_event (srcpad, event);
* gst_object_unref (pad);
* ]|
*
* ## Example launch line
* |[
* gst-launch-1.0 audiotestsrc ! audio/x-raw, channels=4 ! audioconvert mix-matrix="<<(float)1.0, (float)0.0, (float)0.0, (float)0.0>, <(float)0.0, (float)1.0, (float)0.0, (float)0.0>>" ! audio/x-raw,channels=2 ! autoaudiosink
@ -293,6 +303,39 @@ gst_audio_convert_input_channels_reorder_mode_get_type (void)
return reorder_mode_type;
}
static void
gst_audio_convert_set_mix_matrix (GstAudioConvert * this, const GValue * value);
static gboolean
gst_audio_convert_src_event (GstBaseTransform * trans, GstEvent * event)
{
gboolean ret = TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_UPSTREAM:
{
const GstStructure *s = gst_event_get_structure (event);
if (s && gst_structure_has_name (s, "GstRequestAudioMixMatrix")) {
const GValue *matrix = gst_structure_get_value (s, "matrix");
if (matrix) {
gst_audio_convert_set_mix_matrix (GST_AUDIO_CONVERT (trans), matrix);
g_object_notify (G_OBJECT (trans), "mix-matrix");
}
goto done;
}
break;
}
default:
break;
}
ret = GST_BASE_TRANSFORM_CLASS (parent_class)->src_event (trans, event);
done:
return ret;
}
static void
gst_audio_convert_class_init (GstAudioConvertClass * klass)
{
@ -420,6 +463,8 @@ gst_audio_convert_class_init (GstAudioConvertClass * klass)
GST_DEBUG_FUNCPTR (gst_audio_convert_submit_input_buffer);
basetransform_class->prepare_output_buffer =
GST_DEBUG_FUNCPTR (gst_audio_convert_prepare_output_buffer);
basetransform_class->src_event =
GST_DEBUG_FUNCPTR (gst_audio_convert_src_event);
basetransform_class->transform_ip_on_passthrough = FALSE;
@ -1803,7 +1848,7 @@ gst_audio_convert_set_mix_matrix (GstAudioConvert * this, const GValue * value)
{
gboolean mix_matrix_was_set;
GstAudioConverter *old_converter;
GValue old_mix_matrix;
GValue old_mix_matrix = G_VALUE_INIT;
gboolean restore = FALSE;
g_value_init (&old_mix_matrix, GST_TYPE_ARRAY);
@ -1816,9 +1861,7 @@ gst_audio_convert_set_mix_matrix (GstAudioConvert * this, const GValue * value)
g_value_copy (&this->mix_matrix, &old_mix_matrix);
}
if (this->convert) {
this->convert = NULL;
}
this->convert = NULL;
if (!gst_value_array_get_size (value)) {
g_value_copy (value, &this->mix_matrix);
@ -1863,6 +1906,7 @@ done:
} else if (old_converter) {
gst_audio_converter_free (old_converter);
}
g_value_unset (&old_mix_matrix);
}
static void