GstEnginePipeline: rename `audioqueue_` to `inputaudioqueue_`

It is placed in the very beginning, not elsewhere.
This commit is contained in:
Roman Lebedev 2023-08-02 02:22:09 +03:00
parent d2390473bc
commit 1805ebcad3
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GPG Key ID: 083C3EBB4A1689E0
2 changed files with 14 additions and 14 deletions

View File

@ -107,7 +107,7 @@ GstEnginePipeline::GstEnginePipeline(QObject *parent)
pipeline_(nullptr),
audiobin_(nullptr),
audiosink_(nullptr),
audioqueue_(nullptr),
inputaudioqueue_(nullptr),
audioqueueconverter_(nullptr),
ebur128_volume_(nullptr),
volume_(nullptr),
@ -486,8 +486,8 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
// Create all the other elements
audioqueue_ = CreateElement("queue2", "audioqueue", audiobin_, error);
if (!audioqueue_) {
inputaudioqueue_ = CreateElement("queue2", "inputaudioqueue", audiobin_, error);
if (!inputaudioqueue_) {
return false;
}
@ -623,7 +623,7 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
}
{ // Create a pad on the outside of the audiobin and connect it to the pad of the first element.
GstPad *pad = gst_element_get_static_pad(audioqueue_, "sink");
GstPad *pad = gst_element_get_static_pad(inputaudioqueue_, "sink");
if (pad) {
gst_element_add_pad(audiobin_, gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
@ -644,20 +644,20 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
// We set this on this queue instead of the playbin because setting it on the playbin only affects network sources.
// Disable the default buffer and byte limits, so we only buffer based on time.
g_object_set(G_OBJECT(audioqueue_), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(audioqueue_), "max-size-bytes", 0, nullptr);
g_object_set(G_OBJECT(inputaudioqueue_), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(inputaudioqueue_), "max-size-bytes", 0, nullptr);
if (buffer_duration_nanosec_ > 0) {
qLog(Debug) << "Setting buffer duration:" << buffer_duration_nanosec_ << "low watermark:" << buffer_low_watermark_ << "high watermark:" << buffer_high_watermark_;
g_object_set(G_OBJECT(audioqueue_), "use-buffering", true, nullptr);
g_object_set(G_OBJECT(audioqueue_), "max-size-time", buffer_duration_nanosec_, nullptr);
g_object_set(G_OBJECT(audioqueue_), "low-watermark", buffer_low_watermark_, nullptr);
g_object_set(G_OBJECT(audioqueue_), "high-watermark", buffer_high_watermark_, nullptr);
g_object_set(G_OBJECT(inputaudioqueue_), "use-buffering", true, nullptr);
g_object_set(G_OBJECT(inputaudioqueue_), "max-size-time", buffer_duration_nanosec_, nullptr);
g_object_set(G_OBJECT(inputaudioqueue_), "low-watermark", buffer_low_watermark_, nullptr);
g_object_set(G_OBJECT(inputaudioqueue_), "high-watermark", buffer_high_watermark_, nullptr);
}
// Link all elements
if (!gst_element_link(audioqueue_, audioqueueconverter_)) {
error = "Failed to link audio queue to audio queue converter.";
if (!gst_element_link(inputaudioqueue_, audioqueueconverter_)) {
error = "Failed to link input audio queue to audio queue converter.";
return false;
}
@ -1460,7 +1460,7 @@ void GstEnginePipeline::StateChangedMessageReceived(GstMessage *msg) {
void GstEnginePipeline::BufferingMessageReceived(GstMessage *msg) {
// Only handle buffering messages from the queue2 element in audiobin - not the one that's created automatically by playbin.
if (GST_ELEMENT(GST_MESSAGE_SRC(msg)) != audioqueue_) {
if (GST_ELEMENT(GST_MESSAGE_SRC(msg)) != inputaudioqueue_) {
return;
}

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@ -301,7 +301,7 @@ class GstEnginePipeline : public QObject {
GstElement *pipeline_;
GstElement *audiobin_;
GstElement *audiosink_;
GstElement *audioqueue_;
GstElement *inputaudioqueue_;
GstElement *audioqueueconverter_;
GstElement *ebur128_volume_;
GstElement *volume_;