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GstEnginePipeline: Add audio converter for sink

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This commit is contained in:
Jonas Kvinge
2022-12-11 01:14:01 +01:00
parent cc95db25cc
commit c756346357
1 changed files with 21 additions and 7 deletions
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28
src/engine/gstenginepipeline.cpp
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@@ -431,8 +431,15 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
return false; return false;
} }
GstElement *audioconverter = CreateElement("audioconvert", "audioconverter", audiobin_, error); GstElement *audioqueueconverter = CreateElement("audioconvert", "audioqueueconverter", audiobin_, error);
if (!audioconverter) { if (!audioqueueconverter) {
gst_object_unref(GST_OBJECT(audiobin_));
audiobin_ = nullptr;
return false;
}
GstElement *audiosinkconverter = CreateElement("audioconvert", "audiosinkconverter", audiobin_, error);
if (!audiosinkconverter) {
gst_object_unref(GST_OBJECT(audiobin_)); gst_object_unref(GST_OBJECT(audiobin_));
audiobin_ = nullptr; audiobin_ = nullptr;
return false; return false;
@@ -523,7 +530,7 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
} }
// Create the replaygain elements if it's enabled. // Create the replaygain elements if it's enabled.
GstElement *eventprobe = audioconverter; GstElement *eventprobe = audioqueueconverter;
GstElement *rgvolume = nullptr; GstElement *rgvolume = nullptr;
GstElement *rglimiter = nullptr; GstElement *rglimiter = nullptr;
GstElement *rgconverter = nullptr; GstElement *rgconverter = nullptr;
@@ -598,14 +605,14 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
// Link all elements // Link all elements
if (!gst_element_link(audioqueue_, audioconverter)) { if (!gst_element_link(audioqueue_, audioqueueconverter)) {
gst_object_unref(GST_OBJECT(audiobin_)); gst_object_unref(GST_OBJECT(audiobin_));
audiobin_ = nullptr; audiobin_ = nullptr;
error = "gst_element_link() failed."; error = "gst_element_link() failed.";
return false; return false;
} }
GstElement *element_link = audioconverter; // The next element to link from. GstElement *element_link = audioqueueconverter; // The next element to link from.
// Link replaygain elements if enabled. // Link replaygain elements if enabled.
if (rg_enabled_ && rgvolume && rglimiter && rgconverter) { if (rg_enabled_ && rgvolume && rglimiter && rgconverter) {
@@ -674,6 +681,13 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
element_link = bs2b; element_link = bs2b;
} }
if (!gst_element_link(element_link, audiosinkconverter)) {
gst_object_unref(GST_OBJECT(audiobin_));
audiobin_ = nullptr;
error = "gst_element_link() failed.";
return false;
}
{ {
GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw"); GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
if (!caps) { if (!caps) {
@@ -686,12 +700,12 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
qLog(Debug) << "Setting channels to" << channels_; qLog(Debug) << "Setting channels to" << channels_;
gst_caps_set_simple(caps, "channels", G_TYPE_INT, channels_, nullptr); gst_caps_set_simple(caps, "channels", G_TYPE_INT, channels_, nullptr);
} }
gst_element_link_filtered(element_link, audiosink_, caps); gst_element_link_filtered(audiosinkconverter, audiosink_, caps);
gst_caps_unref(caps); gst_caps_unref(caps);
} }
{ // Add probes and handlers. { // Add probes and handlers.
GstPad *pad = gst_element_get_static_pad(audioconverter, "src"); GstPad *pad = gst_element_get_static_pad(audioqueueconverter, "src");
if (pad) { if (pad) {
gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, HandoffCallback, this, nullptr); gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, HandoffCallback, this, nullptr);
gst_object_unref(pad); gst_object_unref(pad);