Funkwhale
/
strawberry-audio-player-windows-mac-linux
mirror of https://github.com/strawberrymusicplayer/strawberry
1
0
Fork 0
Code Issues Releases Wiki Activity

Merge 92f40302ed into e96870cfbd

This commit is contained in:
Roman Lebedev 2023-08-05 02:43:53 +02:00 committed by GitHub
parent e96870cfbd 92f40302ed
commit 3a30a6a90d
No known key found for this signature in database
GPG Key ID: 4AEE18F83AFDEB23
2 changed files with 73 additions and 36 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

105
src/engine/gstenginepipeline.cpp
View File

@ -107,7 +107,8 @@ GstEnginePipeline::GstEnginePipeline(QObject *parent)
pipeline_(nullptr),
audiobin_(nullptr),
audiosink_(nullptr),
audioqueue_(nullptr),
inputaudioqueue_(nullptr),
outputaudioqueue_(nullptr),
audioqueueconverter_(nullptr),
ebur128_volume_(nullptr),
volume_(nullptr),
@ -486,8 +487,8 @@ 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;
}
@ -575,8 +576,6 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
}
eventprobe_ = audioqueueconverter_;
// Create the replaygain elements if it's enabled.
GstElement *rgvolume = nullptr;
GstElement *rglimiter = nullptr;
@ -594,7 +593,6 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
if (!rgconverter) {
return false;
}
eventprobe_ = rgconverter;
// Set replaygain settings
g_object_set(G_OBJECT(rgvolume), "album-mode", rg_mode_, nullptr);
g_object_set(G_OBJECT(rgvolume), "pre-amp", rg_preamp_, nullptr);
@ -611,7 +609,6 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
UpdateEBUR128LoudnessNormalizingGaindB();
eventprobe_ = ebur128_volume_;
}
GstElement *bs2b = nullptr;
@ -622,42 +619,44 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
}
}
outputaudioqueue_ = CreateElement("queue2", "outputaudioqueue", audiobin_, error);
if (!outputaudioqueue_) {
return false;
}
{ // 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 our queues 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.
for(GstElement *audioqueue : {inputaudioqueue_, outputaudioqueue_}) {
g_object_set(G_OBJECT(audioqueue), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(audioqueue), "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);
}
}
// 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.
g_object_set(G_OBJECT(audioqueue_), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(audioqueue_), "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);
}
// We do not care about the buffering status of the input queue,
// only of the output one.
g_object_set(G_OBJECT(inputaudioqueue_), "use-buffering", false, 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;
}
@ -736,6 +735,7 @@ bool GstEnginePipeline::InitAudioBin(QString &error) {
error = "Failed to link audio sink converter.";
return false;
}
element_link = audiosinkconverter;
{
GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
@ -747,17 +747,26 @@ 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, outputaudioqueue_, 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 to output audio queue with filter";
return false;
}
eventprobe_ = outputaudioqueue_;
element_link = outputaudioqueue_;
}
if (!gst_element_link(element_link, audiosink_)) {
error = "Failed to link output audio queue to audio sink for " + output_;
return false;
}
element_link = audiosink_;
{ // Add probes and handlers.
GstPad *pad = gst_element_get_static_pad(audioqueueconverter_, "src");
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);
buffer_probe_cb_id_ = gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, BufferProbeCallback, this, nullptr);
gst_object_unref(pad);
}
@ -1459,8 +1468,12 @@ 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_) {
// Only handle buffering messages from the output queue2 element in audiobin - not the one that's created automatically by playbin.
// We want to monitor the output queue specifically, not the input queue,
// because there are other processing elements inbetween,
// and if the replenishing rate of the *output* queue is below real-time,
// we will certainly have stuttering.
if (GST_ELEMENT(GST_MESSAGE_SRC(msg)) != outputaudioqueue_) {
return;
}
@ -1545,6 +1558,17 @@ bool GstEnginePipeline::Seek(const qint64 nanosec) {
}
static void flush_pipeline(GstElement *pipeline) {
// FIXME: is this really the right way to flush pipeline without seeking?
gint64 position_ns_;
gst_element_query_position(pipeline, GST_FORMAT_TIME, &position_ns_);
gst_element_seek_simple (pipeline, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, position_ns_);
}
QFuture<void> GstEnginePipeline::FlushPipeline() {
return QtConcurrent::run(&set_state_threadpool_, &flush_pipeline, pipeline_);
}
void GstEnginePipeline::SetEBUR128LoudnessNormalizingGain_dB(const double ebur128_loudness_normalizing_gain_db) {
ebur128_loudness_normalizing_gain_db_ = ebur128_loudness_normalizing_gain_db;
@ -1558,6 +1582,8 @@ void GstEnginePipeline::UpdateEBUR128LoudnessNormalizingGaindB() {
auto dB_to_mult = [](const double gain_dB) { return std::pow(10., gain_dB / 20.); };
g_object_set(G_OBJECT(ebur128_volume_), "volume", dB_to_mult(ebur128_loudness_normalizing_gain_db_), nullptr);
FlushPipeline();
}
}
@ -1575,6 +1601,10 @@ void GstEnginePipeline::SetVolume(const uint volume_percent) {
}
}
if (volume_ == volume_sw_) {
FlushPipeline();
}
volume_percent_ = volume_percent;
}
@ -1583,6 +1613,7 @@ void GstEnginePipeline::SetFaderVolume(const qreal volume) {
if (volume_fading_) {
g_object_set(G_OBJECT(volume_fading_), "volume", volume, nullptr);
FlushPipeline(); // FIXME: this is not the right solution.
}
}
@ -1598,6 +1629,8 @@ void GstEnginePipeline::UpdateStereoBalance() {
if (audiopanorama_) {
g_object_set(G_OBJECT(audiopanorama_), "panorama", stereo_balance_, nullptr);
FlushPipeline();
}
}
@ -1637,6 +1670,8 @@ void GstEnginePipeline::UpdateEqualizer() {
g_object_set(G_OBJECT(equalizer_preamp_), "volume", preamp, nullptr);
FlushPipeline();
}
void GstEnginePipeline::StartFader(const qint64 duration_nanosec, const QTimeLine::Direction direction, const QEasingCurve::Type shape, const bool use_fudge_timer) {

4
src/engine/gstenginepipeline.h
View File

@ -85,6 +85,7 @@ class GstEnginePipeline : public QObject {
void RemoveAllBufferConsumers();
// Control the music playback
QFuture<void> FlushPipeline();
QFuture<GstStateChangeReturn> SetState(const GstState state);
Q_INVOKABLE bool Seek(const qint64 nanosec);
void SetEBUR128LoudnessNormalizingGain_dB(const double ebur128_loudness_normalizing_gain_db);
@ -301,7 +302,8 @@ class GstEnginePipeline : public QObject {
GstElement *pipeline_;
GstElement *audiobin_;
GstElement *audiosink_;
GstElement *audioqueue_;
GstElement *inputaudioqueue_;
GstElement *outputaudioqueue_;
GstElement *audioqueueconverter_;
GstElement *ebur128_volume_;
GstElement *volume_;