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GstEnginePipeline: Experiment with input and output queues

This commit is contained in:
Jonas Kvinge 2023-08-03 02:22:53 +02:00
parent c52fc90306
commit 6fc68da1d8
2 changed files with 64 additions and 29 deletions
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89
src/engine/gstenginepipeline.cpp
View File

@ -108,7 +108,8 @@ GstEnginePipeline::GstEnginePipeline(QObject *parent)
pipeline_(nullptr),
audiobin_(nullptr),
audiosink_(nullptr),
audioqueue_(nullptr),
inputaudioqueue_(nullptr),
outputaudioqueue_(nullptr),
audioqueueconverter_(nullptr),
volume_(nullptr),
volume_sw_(nullptr),
@ -118,6 +119,7 @@ GstEnginePipeline::GstEnginePipeline(QObject *parent)
equalizer_(nullptr),
equalizer_preamp_(nullptr),
eventprobe_(nullptr),
bufferprobe_(nullptr),
upstream_events_probe_cb_id_(0),
buffer_probe_cb_id_(0),
playbin_probe_cb_id_(0),
@ -174,7 +176,7 @@ GstEnginePipeline::~GstEnginePipeline() {
}
if (buffer_probe_cb_id_ != 0) {
GstPad *pad = gst_element_get_static_pad(audioqueueconverter_, "src");
GstPad *pad = gst_element_get_static_pad(bufferprobe_, "src");
if (pad) {
gst_pad_remove_probe(pad, buffer_probe_cb_id_);
gst_object_unref(pad);
@ -490,11 +492,18 @@ 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;
}
if (!fading_enabled_) {
outputaudioqueue_ = CreateElement("queue2", "outputaudioqueue", audiobin_, error);
if (!outputaudioqueue_) {
return false;
}
}
audioqueueconverter_ = CreateElement("audioconvert", "audioqueueconverter", audiobin_, error);
if (!audioqueueconverter_) {
return false;
@ -629,39 +638,39 @@ 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);
}
}
// Add a data probe on the src pad of the audioconvert element for our scope.
// We do it here because we want pre-equalized and pre-volume samples so that our visualization are not be affected by them.
{
GstPad *pad = gst_element_get_static_pad(eventprobe_, "src");
if (pad) {
upstream_events_probe_cb_id_ = gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_EVENT_UPSTREAM, &UpstreamEventsProbeCallback, this, nullptr);
gst_object_unref(pad);
}
}
// Set the buffer duration.
// 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.
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-buffers", 0, nullptr);
g_object_set(G_OBJECT(audioqueue_), "max-size-bytes", 0, 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-buffers", 0, nullptr);
g_object_set(G_OBJECT(inputaudioqueue_), "max-size-bytes", 0, 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);
if (outputaudioqueue_) {
g_object_set(G_OBJECT(outputaudioqueue_), "use-buffering", true, nullptr);
g_object_set(G_OBJECT(outputaudioqueue_), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(outputaudioqueue_), "max-size-bytes", 0, nullptr);
g_object_set(G_OBJECT(outputaudioqueue_), "max-size-time", buffer_duration_nanosec_, nullptr);
g_object_set(G_OBJECT(outputaudioqueue_), "low-watermark", buffer_low_watermark_, nullptr);
g_object_set(G_OBJECT(outputaudioqueue_), "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;
}
@ -741,6 +750,9 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
return false;
}
element_link = audiosinkconverter;
GstElement *element_link_filtered = outputaudioqueue_ ? outputaudioqueue_ : audiosink_;
{
GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
if (!caps) {
@ -751,16 +763,37 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
qLog(Debug) << "Setting channels to" << channels_;
gst_caps_set_simple(caps, "channels", G_TYPE_INT, channels_, nullptr);
}
const bool link_filtered_result = gst_element_link_filtered(audiosinkconverter, audiosink_, caps);
const bool link_filtered_result = gst_element_link_filtered(element_link, element_link_filtered, caps);
gst_caps_unref(caps);
if (!link_filtered_result) {
error = "Failed to link audio sink converter to audio sink with filter for " + output_;
error = "Failed to link audio sink converter with filter";
return false;
}
element_link = element_link_filtered;
}
if (element_link != audiosink_) {
if (!gst_element_link(element_link, audiosink_)) {
error = "Failed to link output audio queue to audio sink for " + output_;
return false;
}
}
{ // Add probes and handlers.
GstPad *pad = gst_element_get_static_pad(audioqueueconverter_, "src");
if (outputaudioqueue_) {
eventprobe_ = outputaudioqueue_;
}
{
GstPad *pad = gst_element_get_static_pad(eventprobe_, "src");
if (pad) {
upstream_events_probe_cb_id_ = gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_EVENT_UPSTREAM, &UpstreamEventsProbeCallback, this, nullptr);
gst_object_unref(pad);
}
}
{
bufferprobe_ = outputaudioqueue_ ? outputaudioqueue_ : audiosinkconverter;
GstPad *pad = gst_element_get_static_pad(bufferprobe_, "src");
if (pad) {
buffer_probe_cb_id_ = gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, BufferProbeCallback, this, nullptr);
gst_object_unref(pad);
@ -1483,7 +1516,7 @@ 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)) != outputaudioqueue_) {
return;
}

4
src/engine/gstenginepipeline.h
View File

@ -306,7 +306,8 @@ class GstEnginePipeline : public QObject {
GstElement *pipeline_;
GstElement *audiobin_;
GstElement *audiosink_;
GstElement *audioqueue_;
GstElement *inputaudioqueue_;
GstElement *outputaudioqueue_;
GstElement *audioqueueconverter_;
GstElement *volume_;
GstElement *volume_sw_;
@ -316,6 +317,7 @@ class GstEnginePipeline : public QObject {
GstElement *equalizer_;
GstElement *equalizer_preamp_;
GstElement *eventprobe_;
GstElement *bufferprobe_;
gulong upstream_events_probe_cb_id_;
gulong buffer_probe_cb_id_;