Refactor gstreamer engine code, equalizer and fix stereo balancer
This commit is contained in:
parent
d033b79af4
commit
834877c503
@ -200,12 +200,13 @@ void Player::Init() {
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connect(engine_.get(), SIGNAL(MetaData(Engine::SimpleMetaBundle)), SLOT(EngineMetadataReceived(Engine::SimpleMetaBundle)));
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// Equalizer
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qLog(Debug) << "Creating equalizer";
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connect(equalizer_, SIGNAL(ParametersChanged(int,QList<int>)), app_->player()->engine(), SLOT(SetEqualizerParameters(int,QList<int>)));
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connect(equalizer_, SIGNAL(EnabledChanged(bool)), app_->player()->engine(), SLOT(SetEqualizerEnabled(bool)));
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connect(equalizer_, SIGNAL(StereoBalanceChanged(bool, float)), app_->player()->engine(), SLOT(SetStereoBalance(bool, float)));
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connect(equalizer_, SIGNAL(StereoBalancerEnabledChanged(bool)), app_->player()->engine(), SLOT(SetStereoBalancerEnabled(bool)));
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connect(equalizer_, SIGNAL(StereoBalanceChanged(float)), app_->player()->engine(), SLOT(SetStereoBalance(float)));
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connect(equalizer_, SIGNAL(EqualizerEnabledChanged(bool)), app_->player()->engine(), SLOT(SetEqualizerEnabled(bool)));
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connect(equalizer_, SIGNAL(EqualizerParametersChanged(int, QList<int>)), app_->player()->engine(), SLOT(SetEqualizerParameters(int, QList<int>)));
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engine_->SetStereoBalance(equalizer_->is_stereo_balancer_enabled(), equalizer_->stereo_balance());
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engine_->SetStereoBalancerEnabled(equalizer_->is_stereo_balancer_enabled());
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engine_->SetStereoBalance(equalizer_->stereo_balance());
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engine_->SetEqualizerEnabled(equalizer_->is_equalizer_enabled());
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engine_->SetEqualizerParameters(equalizer_->preamp_value(), equalizer_->gain_values());
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@ -82,7 +82,7 @@ bool Engine::Base::Play(const QUrl &stream_url, const QUrl &original_url, TrackC
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}
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void Engine::Base::SetVolume(uint value) {
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void Engine::Base::SetVolume(const uint value) {
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volume_ = value;
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SetVolumeSW(MakeVolumeLogarithmic(value));
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@ -126,9 +126,10 @@ public:
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QVariant device() { return device_; }
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public slots:
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virtual void SetStereoBalancerEnabled(const bool) {}
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virtual void SetStereoBalance(const float) {}
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virtual void SetEqualizerEnabled(const bool) {}
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virtual void SetEqualizerParameters(const int preamp, const QList<int> &bandGains) { Q_UNUSED(preamp); Q_UNUSED(bandGains); }
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virtual void SetStereoBalance(const bool enabled, const float value) { Q_UNUSED(enabled); Q_UNUSED(value); }
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virtual void SetEqualizerParameters(const int, const QList<int>&) {}
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signals:
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// Emitted when crossfading is enabled and the track is crossfade_duration_ away from finishing
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@ -85,6 +85,8 @@ GstEngine::GstEngine(TaskManager *task_manager)
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latest_buffer_(nullptr),
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stereo_balancer_enabled_(false),
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stereo_balance_(0.0f),
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equalizer_enabled_(false),
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equalizer_preamp_(0),
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seek_timer_(new QTimer(this)),
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timer_id_(-1),
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next_element_id_(0),
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@ -103,12 +105,15 @@ GstEngine::GstEngine(TaskManager *task_manager)
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}
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GstEngine::~GstEngine() {
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EnsureInitialised();
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current_pipeline_.reset();
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if (latest_buffer_) {
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gst_buffer_unref(latest_buffer_);
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latest_buffer_ = nullptr;
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}
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}
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bool GstEngine::Init() {
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@ -175,7 +180,7 @@ bool GstEngine::Load(const QUrl &stream_url, const QUrl &original_url, Engine::T
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current_pipeline_ = pipeline;
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SetVolume(volume_);
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SetStereoBalance(stereo_balancer_enabled_, stereo_balance_);
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SetStereoBalance(stereo_balance_);
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SetEqualizerParameters(equalizer_preamp_, equalizer_gains_);
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// Maybe fade in this track
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@ -425,11 +430,25 @@ void GstEngine::ConsumeBuffer(GstBuffer *buffer, const int pipeline_id, const QS
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}
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void GstEngine::SetStereoBalancerEnabled(const bool enabled) {
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stereo_balancer_enabled_ = enabled;
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if (current_pipeline_) current_pipeline_->set_stereo_balancer_enabled(enabled);
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}
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void GstEngine::SetStereoBalance(const float value) {
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stereo_balance_ = value;
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if (current_pipeline_) current_pipeline_->SetStereoBalance(value);
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}
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void GstEngine::SetEqualizerEnabled(const bool enabled) {
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equalizer_enabled_ = enabled;
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if (current_pipeline_) current_pipeline_->set_equalizer_enabled(enabled);
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if (current_pipeline_) current_pipeline_->SetEqualizerEnabled(enabled);
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}
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void GstEngine::SetEqualizerParameters(const int preamp, const QList<int> &band_gains) {
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@ -437,27 +456,22 @@ void GstEngine::SetEqualizerParameters(const int preamp, const QList<int> &band_
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equalizer_preamp_ = preamp;
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equalizer_gains_ = band_gains;
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if (current_pipeline_)
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current_pipeline_->SetEqualizerParams(preamp, band_gains);
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}
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void GstEngine::SetStereoBalance(const bool enabled, const float value) {
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stereo_balance_ = value;
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if (current_pipeline_) current_pipeline_->SetStereoBalance(enabled, value);
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if (current_pipeline_) current_pipeline_->SetEqualizerParams(preamp, band_gains);
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}
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void GstEngine::AddBufferConsumer(GstBufferConsumer *consumer) {
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buffer_consumers_ << consumer;
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if (current_pipeline_) current_pipeline_->AddBufferConsumer(consumer);
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}
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void GstEngine::RemoveBufferConsumer(GstBufferConsumer *consumer) {
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buffer_consumers_.removeAll(consumer);
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if (current_pipeline_) current_pipeline_->RemoveBufferConsumer(consumer);
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}
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void GstEngine::timerEvent(QTimerEvent *e) {
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@ -745,11 +759,12 @@ shared_ptr<GstEnginePipeline> GstEngine::CreatePipeline() {
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shared_ptr<GstEnginePipeline> ret(new GstEnginePipeline(this));
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ret->set_output_device(output_, device_);
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ret->set_volume_control(volume_control_);
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ret->set_volume_enabled(volume_control_);
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ret->set_stereo_balancer_enabled(stereo_balancer_enabled_);
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ret->set_equalizer_enabled(equalizer_enabled_);
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ret->set_replaygain(rg_enabled_, rg_mode_, rg_preamp_, rg_compression_);
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ret->set_buffer_duration_nanosec(buffer_duration_nanosec_);
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ret->set_buffer_min_fill(buffer_min_fill_);
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ret->SetEqualizerEnabled(equalizer_enabled_);
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ret->AddBufferConsumer(this);
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for (GstBufferConsumer *consumer : buffer_consumers_) {
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@ -95,15 +95,18 @@ class GstEngine : public Engine::Base, public GstBufferConsumer {
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public slots:
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void ReloadSettings();
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/** Set whether equalizer is enabled */
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// Set whether stereo balancer is enabled
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void SetStereoBalancerEnabled(const bool enabled);
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// Set Stereo balance, range -1.0f..1.0f
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void SetStereoBalance(const float value);
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// Set whether equalizer is enabled
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void SetEqualizerEnabled(const bool);
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/** Set equalizer preamp and gains, range -100..100. Gains are 10 values. */
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// Set equalizer preamp and gains, range -100..100. Gains are 10 values.
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void SetEqualizerParameters(const int preamp, const QList<int> &bandGains);
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/** Set Stereo balance, range -1.0f..1.0f */
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void SetStereoBalance(const bool enabled, const float value);
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void AddBufferConsumer(GstBufferConsumer *consumer);
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void RemoveBufferConsumer(GstBufferConsumer *consumer);
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@ -173,6 +176,7 @@ class GstEngine : public Engine::Base, public GstBufferConsumer {
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bool stereo_balancer_enabled_;
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float stereo_balance_;
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bool equalizer_enabled_;
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int equalizer_preamp_;
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QList<int> equalizer_gains_;
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@ -67,12 +67,12 @@ GstEnginePipeline::GstEnginePipeline(GstEngine *engine)
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engine_(engine),
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id_(sId++),
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valid_(false),
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volume_control_(true),
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stereo_balance_enabled_(false),
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stereo_balance_(0.0f),
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volume_enabled_(true),
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stereo_balancer_enabled_(false),
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eq_enabled_(false),
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eq_preamp_(0),
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rg_enabled_(false),
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stereo_balance_(0.0f),
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eq_preamp_(0),
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rg_mode_(0),
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rg_preamp_(0.0),
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rg_compression_(true),
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@ -89,9 +89,10 @@ GstEnginePipeline::GstEnginePipeline(GstEngine *engine)
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pipeline_is_initialised_(false),
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pipeline_is_connected_(false),
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pending_seek_nanosec_(-1),
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last_known_position_ns_(0),
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next_uri_set_(false),
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volume_percent_(100),
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volume_modifier_(1.0),
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volume_modifier_(1.0f),
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use_fudge_timer_(false),
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pipeline_(nullptr),
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audiobin_(nullptr),
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@ -101,9 +102,9 @@ GstEnginePipeline::GstEnginePipeline(GstEngine *engine)
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equalizer_(nullptr),
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equalizer_preamp_(nullptr),
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discoverer_(nullptr),
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about_to_finish_cb_id_(-1),
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pad_added_cb_id_(-1),
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notify_source_cb_id_(-1),
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about_to_finish_cb_id_(-1),
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bus_cb_id_(-1),
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discovery_finished_cb_id_(-1),
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discovery_discovered_cb_id_(-1)
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@ -131,15 +132,15 @@ GstEnginePipeline::~GstEnginePipeline() {
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if (pipeline_) {
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if (about_to_finish_cb_id_ != -1)
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g_signal_handler_disconnect(G_OBJECT(pipeline_), about_to_finish_cb_id_);
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if (pad_added_cb_id_ != -1)
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g_signal_handler_disconnect(G_OBJECT(pipeline_), pad_added_cb_id_);
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if (notify_source_cb_id_ != -1)
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g_signal_handler_disconnect(G_OBJECT(pipeline_), notify_source_cb_id_);
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if (about_to_finish_cb_id_ != -1)
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g_signal_handler_disconnect(G_OBJECT(pipeline_), about_to_finish_cb_id_);
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gst_bus_set_sync_handler(gst_pipeline_get_bus(GST_PIPELINE(pipeline_)), nullptr, nullptr, nullptr);
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if (bus_cb_id_ != -1)
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@ -159,13 +160,22 @@ void GstEnginePipeline::set_output_device(const QString &output, const QVariant
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}
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void GstEnginePipeline::set_volume_control(bool volume_control) {
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volume_control_ = volume_control;
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void GstEnginePipeline::set_volume_enabled(const bool enabled) {
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volume_enabled_ = enabled;
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}
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void GstEnginePipeline::set_replaygain(bool enabled, int mode, float preamp, bool compression) {
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void GstEnginePipeline::set_stereo_balancer_enabled(const bool enabled) {
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stereo_balancer_enabled_ = enabled;
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if (!enabled) stereo_balance_ = 0.0f;
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if (pipeline_) UpdateStereoBalance();
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}
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void GstEnginePipeline::set_equalizer_enabled(const bool enabled) {
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eq_enabled_ = enabled;
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if (pipeline_) UpdateEqualizer();
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}
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void GstEnginePipeline::set_replaygain(const bool enabled, const int mode, const float preamp, const bool compression) {
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rg_enabled_ = enabled;
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rg_mode_ = mode;
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@ -182,6 +192,45 @@ void GstEnginePipeline::set_buffer_min_fill(int percent) {
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buffer_min_fill_ = percent;
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}
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bool GstEnginePipeline::InitFromUrl(const QByteArray &stream_url, const QUrl original_url, const qint64 end_nanosec) {
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stream_url_ = stream_url;
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original_url_ = original_url;
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end_offset_nanosec_ = end_nanosec;
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pipeline_ = engine_->CreateElement("playbin");
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if (!pipeline_) return false;
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g_object_set(G_OBJECT(pipeline_), "uri", stream_url.constData(), nullptr);
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gint flags;
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g_object_get(G_OBJECT(pipeline_), "flags", &flags, nullptr);
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flags |= 0x00000002;
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flags &= ~0x00000001;
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g_object_set(G_OBJECT(pipeline_), "flags", flags, nullptr);
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pad_added_cb_id_ = CHECKED_GCONNECT(G_OBJECT(pipeline_), "pad-added", &NewPadCallback, this);
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notify_source_cb_id_ = CHECKED_GCONNECT(G_OBJECT(pipeline_), "notify::source", &SourceSetupCallback, this);
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about_to_finish_cb_id_ = CHECKED_GCONNECT(G_OBJECT(pipeline_), "about-to-finish", &AboutToFinishCallback, this);
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// Setting up a discoverer
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discoverer_ = gst_discoverer_new(kDiscoveryTimeoutS * GST_SECOND, nullptr);
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if (discoverer_) {
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discovery_discovered_cb_id_ = CHECKED_GCONNECT(G_OBJECT(discoverer_), "discovered", &StreamDiscovered, this);
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discovery_finished_cb_id_ = CHECKED_GCONNECT(G_OBJECT(discoverer_), "finished", &StreamDiscoveryFinished, this);
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gst_discoverer_start(discoverer_);
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}
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if (!InitAudioBin()) return false;
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// Set playbin's sink to be our custom audio-sink.
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g_object_set(GST_OBJECT(pipeline_), "audio-sink", audiobin_, nullptr);
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pipeline_is_connected_ = true;
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return true;
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}
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bool GstEnginePipeline::InitAudioBin() {
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gst_segment_init(&last_playbin_segment_, GST_FORMAT_TIME);
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@ -239,12 +288,12 @@ bool GstEnginePipeline::InitAudioBin() {
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}
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// Create the volume elements if it's enabled.
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if (volume_control_) {
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if (volume_enabled_) {
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volume_ = engine_->CreateElement("volume", audiobin_);
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}
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// Create the stereo balancer elements if it's enabled.
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if (stereo_balance_enabled_) {
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if (stereo_balancer_enabled_) {
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audiopanorama_ = engine_->CreateElement("audiopanorama", audiobin_, false);
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// Set the stereo balance.
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if (audiopanorama_) g_object_set(G_OBJECT(audiopanorama_), "panorama", stereo_balance_, nullptr);
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@ -293,13 +342,13 @@ bool GstEnginePipeline::InitAudioBin() {
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GstElement *eventprobe = audioqueue_;
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GstElement *rgvolume = nullptr;
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GstElement *rglimiter = nullptr;
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GstElement *audioconverter2 = nullptr;
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GstElement *rgconverter = nullptr;
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if (rg_enabled_) {
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rgvolume = engine_->CreateElement("rgvolume", audiobin_, false);
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rglimiter = engine_->CreateElement("rglimiter", audiobin_, false);
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audioconverter2 = engine_->CreateElement("audioconvert", audiobin_, false);
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if (rgvolume && rglimiter && audioconverter2) {
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eventprobe = audioconverter2;
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rgconverter = engine_->CreateElement("audioconvert", audiobin_, false);
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if (rgvolume && rglimiter && rgconverter) {
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eventprobe = rgconverter;
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// Set replaygain settings
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g_object_set(G_OBJECT(rgvolume), "album-mode", rg_mode_, nullptr);
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g_object_set(G_OBJECT(rgvolume), "pre-amp", double(rg_preamp_), nullptr);
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@ -335,9 +384,9 @@ bool GstEnginePipeline::InitAudioBin() {
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GstElement *next = audioqueue_; // The next element to link from.
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// Link replaygain elements if enabled.
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if (rg_enabled_ && rgvolume && rglimiter && audioconverter2) {
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gst_element_link_many(next, rgvolume, rglimiter, audioconverter2, nullptr);
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next = audioconverter2;
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if (rg_enabled_ && rgvolume && rglimiter && rgconverter) {
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gst_element_link_many(next, rgvolume, rglimiter, rgconverter, nullptr);
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next = rgconverter;
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}
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// Link equalizer elements if enabled.
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@ -347,20 +396,19 @@ bool GstEnginePipeline::InitAudioBin() {
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}
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// Link equalizer elements if enabled.
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if (stereo_balance_enabled_ && audiopanorama_) {
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if (stereo_balancer_enabled_ && audiopanorama_) {
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gst_element_link(next, audiopanorama_);
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next = audiopanorama_;
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}
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// Link volume elements if enabled.
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if (volume_control_ && volume_) {
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if (volume_enabled_ && volume_) {
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gst_element_link(next, volume_);
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next = volume_;
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}
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gst_element_link(next, audioconverter);
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// Let the audio output of the tee autonegotiate the bit depth and format.
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GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
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gst_element_link_filtered(audioconverter, audiosink, caps);
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gst_caps_unref(caps);
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@ -386,42 +434,225 @@ bool GstEnginePipeline::InitAudioBin() {
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}
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bool GstEnginePipeline::InitFromUrl(const QByteArray &stream_url, const QUrl original_url, const qint64 end_nanosec) {
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GstPadProbeReturn GstEnginePipeline::EventHandoffCallback(GstPad*, GstPadProbeInfo *info, gpointer self) {
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stream_url_ = stream_url;
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original_url_ = original_url;
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end_offset_nanosec_ = end_nanosec;
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GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
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pipeline_ = engine_->CreateElement("playbin");
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if (!pipeline_) return false;
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GstEvent *e = gst_pad_probe_info_get_event(info);
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g_object_set(G_OBJECT(pipeline_), "uri", stream_url.constData(), nullptr);
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qLog(Debug) << instance->id() << "event" << GST_EVENT_TYPE_NAME(e);
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gint flags;
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g_object_get(G_OBJECT(pipeline_), "flags", &flags, nullptr);
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flags |= 0x00000002;
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flags &= ~0x00000001;
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g_object_set(G_OBJECT(pipeline_), "flags", flags, nullptr);
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switch (GST_EVENT_TYPE(e)) {
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case GST_EVENT_SEGMENT:
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if (!instance->segment_start_received_) {
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// The segment start time is used to calculate the proper offset of data buffers from the start of the stream
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const GstSegment *segment = nullptr;
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gst_event_parse_segment(e, &segment);
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instance->segment_start_ = segment->start;
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instance->segment_start_received_ = true;
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}
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break;
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||||
|
||||
about_to_finish_cb_id_ = CHECKED_GCONNECT(G_OBJECT(pipeline_), "about-to-finish", &AboutToFinishCallback, this);
|
||||
pad_added_cb_id_ = CHECKED_GCONNECT(G_OBJECT(pipeline_), "pad-added", &NewPadCallback, this);
|
||||
notify_source_cb_id_ = CHECKED_GCONNECT(G_OBJECT(pipeline_), "notify::source", &SourceSetupCallback, this);
|
||||
|
||||
// Setting up a discoverer
|
||||
discoverer_ = gst_discoverer_new(kDiscoveryTimeoutS * GST_SECOND, nullptr);
|
||||
if (discoverer_) {
|
||||
discovery_discovered_cb_id_ = CHECKED_GCONNECT(G_OBJECT(discoverer_), "discovered", &StreamDiscovered, this);
|
||||
discovery_finished_cb_id_ = CHECKED_GCONNECT(G_OBJECT(discoverer_), "finished", &StreamDiscoveryFinished, this);
|
||||
gst_discoverer_start(discoverer_);
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
if (!InitAudioBin()) return false;
|
||||
return GST_PAD_PROBE_OK;
|
||||
|
||||
// Set playbin's sink to be our costum audio-sink.
|
||||
g_object_set(GST_OBJECT(pipeline_), "audio-sink", audiobin_, nullptr);
|
||||
pipeline_is_connected_ = true;
|
||||
}
|
||||
|
||||
return true;
|
||||
void GstEnginePipeline::SourceSetupCallback(GstPlayBin *bin, GParamSpec *, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
|
||||
GstElement *element = nullptr;
|
||||
g_object_get(bin, "source", &element, nullptr);
|
||||
if (!element) {
|
||||
return;
|
||||
}
|
||||
|
||||
if (g_object_class_find_property(G_OBJECT_GET_CLASS(element), "device") && !instance->source_device().isEmpty()) {
|
||||
// Gstreamer is not able to handle device in URL (referring to Gstreamer documentation, this might be added in the future).
|
||||
// Despite that, for now we include device inside URL: we decompose it during Init and set device here, when this callback is called.
|
||||
g_object_set(element, "device", instance->source_device().toLocal8Bit().constData(), nullptr);
|
||||
}
|
||||
|
||||
if (g_object_class_find_property(G_OBJECT_GET_CLASS(element), "user-agent")) {
|
||||
QString user_agent = QString("%1 %2").arg(QCoreApplication::applicationName(), QCoreApplication::applicationVersion());
|
||||
g_object_set(element, "user-agent", user_agent.toUtf8().constData(), nullptr);
|
||||
g_object_set(element, "ssl-strict", FALSE, nullptr);
|
||||
}
|
||||
|
||||
// If the pipeline was buffering we stop that now.
|
||||
if (instance->buffering_) {
|
||||
instance->buffering_ = false;
|
||||
emit instance->BufferingFinished();
|
||||
instance->SetState(GST_STATE_PLAYING);
|
||||
}
|
||||
|
||||
g_object_unref(element);
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::NewPadCallback(GstElement*, GstPad *pad, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
|
||||
GstPad *const audiopad = gst_element_get_static_pad(instance->audiobin_, "sink");
|
||||
|
||||
// Link playbin's sink pad to audiobin's src pad.
|
||||
if (GST_PAD_IS_LINKED(audiopad)) {
|
||||
qLog(Warning) << instance->id() << "audiopad is already linked, unlinking old pad";
|
||||
gst_pad_unlink(audiopad, GST_PAD_PEER(audiopad));
|
||||
}
|
||||
|
||||
gst_pad_link(pad, audiopad);
|
||||
gst_object_unref(audiopad);
|
||||
|
||||
// Offset the timestamps on all the buffers coming out of the playbin so they line up exactly with the end of the last buffer from the old playbin.
|
||||
// "Running time" is the time since the last flushing seek.
|
||||
GstClockTime running_time = gst_segment_to_running_time(&instance->last_playbin_segment_, GST_FORMAT_TIME, instance->last_playbin_segment_.position);
|
||||
gst_pad_set_offset(pad, running_time);
|
||||
|
||||
// Add a probe to the pad so we can update last_playbin_segment_.
|
||||
gst_pad_add_probe(pad, static_cast<GstPadProbeType>(GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM | GST_PAD_PROBE_TYPE_EVENT_FLUSH), PlaybinProbe, instance, nullptr);
|
||||
|
||||
instance->pipeline_is_connected_ = true;
|
||||
if (instance->pending_seek_nanosec_ != -1 && instance->pipeline_is_initialised_) {
|
||||
QMetaObject::invokeMethod(instance, "Seek", Qt::QueuedConnection, Q_ARG(qint64, instance->pending_seek_nanosec_));
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
GstPadProbeReturn GstEnginePipeline::PlaybinProbe(GstPad *pad, GstPadProbeInfo *info, gpointer data) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(data);
|
||||
|
||||
const GstPadProbeType info_type = GST_PAD_PROBE_INFO_TYPE(info);
|
||||
|
||||
if (info_type & GST_PAD_PROBE_TYPE_BUFFER) {
|
||||
// The playbin produced a buffer. Record its end time, so we can offset the buffers produced by the next playbin when transitioning to the next song.
|
||||
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER(info);
|
||||
|
||||
GstClockTime timestamp = GST_BUFFER_TIMESTAMP(buffer);
|
||||
GstClockTime duration = GST_BUFFER_DURATION(buffer);
|
||||
if (timestamp == GST_CLOCK_TIME_NONE) {
|
||||
timestamp = instance->last_playbin_segment_.position;
|
||||
}
|
||||
|
||||
if (duration != GST_CLOCK_TIME_NONE) {
|
||||
timestamp += duration;
|
||||
}
|
||||
|
||||
instance->last_playbin_segment_.position = timestamp;
|
||||
}
|
||||
else if (info_type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) {
|
||||
GstEvent *event = GST_PAD_PROBE_INFO_EVENT(info);
|
||||
GstEventType event_type = GST_EVENT_TYPE(event);
|
||||
|
||||
if (event_type == GST_EVENT_SEGMENT) {
|
||||
// A new segment started, we need to save this to calculate running time offsets later.
|
||||
gst_event_copy_segment(event, &instance->last_playbin_segment_);
|
||||
}
|
||||
else if (event_type == GST_EVENT_FLUSH_START) {
|
||||
// A flushing seek resets the running time to 0, so remove any offset we set on this pad before.
|
||||
gst_pad_set_offset(pad, 0);
|
||||
}
|
||||
}
|
||||
|
||||
return GST_PAD_PROBE_OK;
|
||||
|
||||
}
|
||||
|
||||
GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad *pad, GstPadProbeInfo *info, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
|
||||
GstCaps *caps = gst_pad_get_current_caps(pad);
|
||||
GstStructure *structure = gst_caps_get_structure(caps, 0);
|
||||
QString format = QString(gst_structure_get_string(structure, "format"));
|
||||
int channels = 0;
|
||||
int rate = 0;
|
||||
gst_structure_get_int(structure, "channels", &channels);
|
||||
gst_structure_get_int(structure, "rate", &rate);
|
||||
|
||||
GstBuffer *buf = gst_pad_probe_info_get_buffer(info);
|
||||
GstBuffer *buf16 = nullptr;
|
||||
|
||||
if (format.startsWith("S32")) {
|
||||
|
||||
GstMapInfo map_info;
|
||||
gst_buffer_map(buf, &map_info, GST_MAP_READ);
|
||||
|
||||
int32_t *s = (int32_t*) map_info.data;
|
||||
int samples = (map_info.size / sizeof(int32_t)) / channels;
|
||||
int buf16_size = samples * sizeof(int16_t) * channels;
|
||||
int16_t *d = (int16_t*) g_malloc(buf16_size);
|
||||
memset(d, 0, buf16_size);
|
||||
for (int i = 0 ; i < (samples * 2) ; ++i) {
|
||||
d[i] = (int16_t) (s[i] >> 16);
|
||||
}
|
||||
gst_buffer_unmap(buf, &map_info);
|
||||
buf16 = gst_buffer_new_wrapped(d, buf16_size);
|
||||
GST_BUFFER_DURATION(buf16) = GST_FRAMES_TO_CLOCK_TIME(samples * sizeof(int16_t) * channels, rate);
|
||||
buf = buf16;
|
||||
}
|
||||
|
||||
QList<GstBufferConsumer*> consumers;
|
||||
{
|
||||
QMutexLocker l(&instance->buffer_consumers_mutex_);
|
||||
consumers = instance->buffer_consumers_;
|
||||
}
|
||||
|
||||
for (GstBufferConsumer *consumer : consumers) {
|
||||
gst_buffer_ref(buf);
|
||||
consumer->ConsumeBuffer(buf, instance->id(), format);
|
||||
}
|
||||
|
||||
if (buf16) {
|
||||
gst_buffer_unref(buf16);
|
||||
}
|
||||
|
||||
// Calculate the end time of this buffer so we can stop playback if it's after the end time of this song.
|
||||
if (instance->end_offset_nanosec_ > 0) {
|
||||
quint64 start_time = GST_BUFFER_TIMESTAMP(buf) - instance->segment_start_;
|
||||
quint64 duration = GST_BUFFER_DURATION(buf);
|
||||
quint64 end_time = start_time + duration;
|
||||
|
||||
if (end_time > instance->end_offset_nanosec_) {
|
||||
if (instance->has_next_valid_url() && instance->next_stream_url_ == instance->stream_url_ && instance->next_beginning_offset_nanosec_ == instance->end_offset_nanosec_) {
|
||||
// The "next" song is actually the next segment of this file - so cheat and keep on playing, but just tell the Engine we've moved on.
|
||||
instance->end_offset_nanosec_ = instance->next_end_offset_nanosec_;
|
||||
instance->next_stream_url_.clear();
|
||||
instance->next_original_url_.clear();
|
||||
instance->next_beginning_offset_nanosec_ = 0;
|
||||
instance->next_end_offset_nanosec_ = 0;
|
||||
|
||||
// GstEngine will try to seek to the start of the new section, but we're already there so ignore it.
|
||||
instance->ignore_next_seek_ = true;
|
||||
emit instance->EndOfStreamReached(instance->id(), true);
|
||||
}
|
||||
else {
|
||||
// There's no next song
|
||||
emit instance->EndOfStreamReached(instance->id(), false);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
return GST_PAD_PROBE_OK;
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::AboutToFinishCallback(GstPlayBin*, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
|
||||
if (instance->has_next_valid_url() && !instance->next_uri_set_) {
|
||||
// Set the next uri. When the current song ends it will be played automatically and a STREAM_START message is send to the bus.
|
||||
// When the next uri is not playable an error message is send when the pipeline goes to PLAY (or PAUSE) state or immediately if it is currently in PLAY state.
|
||||
instance->next_uri_set_ = true;
|
||||
g_object_set(G_OBJECT(instance->pipeline_), "uri", instance->next_stream_url_.constData(), nullptr);
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
@ -710,230 +941,8 @@ void GstEnginePipeline::BufferingMessageReceived(GstMessage *msg) {
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::NewPadCallback(GstElement*, GstPad *pad, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
if (!instance) return;
|
||||
|
||||
GstPad *const audiopad = gst_element_get_static_pad(instance->audiobin_, "sink");
|
||||
|
||||
// Link playbin's sink pad to audiobin's src pad.
|
||||
if (GST_PAD_IS_LINKED(audiopad)) {
|
||||
qLog(Warning) << instance->id() << "audiopad is already linked, unlinking old pad";
|
||||
gst_pad_unlink(audiopad, GST_PAD_PEER(audiopad));
|
||||
}
|
||||
|
||||
gst_pad_link(pad, audiopad);
|
||||
gst_object_unref(audiopad);
|
||||
|
||||
// Offset the timestamps on all the buffers coming out of the playbin so they line up exactly with the end of the last buffer from the old playbin.
|
||||
// "Running time" is the time since the last flushing seek.
|
||||
GstClockTime running_time = gst_segment_to_running_time(&instance->last_playbin_segment_, GST_FORMAT_TIME, instance->last_playbin_segment_.position);
|
||||
gst_pad_set_offset(pad, running_time);
|
||||
|
||||
// Add a probe to the pad so we can update last_playbin_segment_.
|
||||
gst_pad_add_probe(pad, static_cast<GstPadProbeType>(GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM | GST_PAD_PROBE_TYPE_EVENT_FLUSH), PlaybinProbe, instance, nullptr);
|
||||
|
||||
instance->pipeline_is_connected_ = true;
|
||||
if (instance->pending_seek_nanosec_ != -1 && instance->pipeline_is_initialised_) {
|
||||
QMetaObject::invokeMethod(instance, "Seek", Qt::QueuedConnection, Q_ARG(qint64, instance->pending_seek_nanosec_));
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
GstPadProbeReturn GstEnginePipeline::PlaybinProbe(GstPad *pad, GstPadProbeInfo *info, gpointer data) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(data);
|
||||
|
||||
const GstPadProbeType info_type = GST_PAD_PROBE_INFO_TYPE(info);
|
||||
|
||||
if (info_type & GST_PAD_PROBE_TYPE_BUFFER) {
|
||||
// The playbin produced a buffer. Record its end time, so we can offset the buffers produced by the next playbin when transitioning to the next song.
|
||||
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER(info);
|
||||
|
||||
GstClockTime timestamp = GST_BUFFER_TIMESTAMP(buffer);
|
||||
GstClockTime duration = GST_BUFFER_DURATION(buffer);
|
||||
if (timestamp == GST_CLOCK_TIME_NONE) {
|
||||
timestamp = instance->last_playbin_segment_.position;
|
||||
}
|
||||
|
||||
if (duration != GST_CLOCK_TIME_NONE) {
|
||||
timestamp += duration;
|
||||
}
|
||||
|
||||
instance->last_playbin_segment_.position = timestamp;
|
||||
}
|
||||
else if (info_type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) {
|
||||
GstEvent *event = GST_PAD_PROBE_INFO_EVENT(info);
|
||||
GstEventType event_type = GST_EVENT_TYPE(event);
|
||||
|
||||
if (event_type == GST_EVENT_SEGMENT) {
|
||||
// A new segment started, we need to save this to calculate running time offsets later.
|
||||
gst_event_copy_segment(event, &instance->last_playbin_segment_);
|
||||
}
|
||||
else if (event_type == GST_EVENT_FLUSH_START) {
|
||||
// A flushing seek resets the running time to 0, so remove any offset we set on this pad before.
|
||||
gst_pad_set_offset(pad, 0);
|
||||
}
|
||||
}
|
||||
|
||||
return GST_PAD_PROBE_OK;
|
||||
|
||||
}
|
||||
|
||||
GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad *pad, GstPadProbeInfo *info, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
|
||||
GstCaps *caps = gst_pad_get_current_caps(pad);
|
||||
GstStructure *structure = gst_caps_get_structure(caps, 0);
|
||||
QString format = QString(gst_structure_get_string(structure, "format"));
|
||||
int channels = 0;
|
||||
int rate = 0;
|
||||
gst_structure_get_int(structure, "channels", &channels);
|
||||
gst_structure_get_int(structure, "rate", &rate);
|
||||
|
||||
GstBuffer *buf = gst_pad_probe_info_get_buffer(info);
|
||||
GstBuffer *buf16 = nullptr;
|
||||
|
||||
if (format.startsWith("S32")) {
|
||||
|
||||
GstMapInfo map_info;
|
||||
gst_buffer_map(buf, &map_info, GST_MAP_READ);
|
||||
|
||||
int32_t *s = (int32_t*) map_info.data;
|
||||
int samples = (map_info.size / sizeof(int32_t)) / channels;
|
||||
int buf16_size = samples * sizeof(int16_t) * channels;
|
||||
int16_t *d = (int16_t*) g_malloc(buf16_size);
|
||||
memset(d, 0, buf16_size);
|
||||
for (int i = 0 ; i < (samples * 2) ; ++i) {
|
||||
d[i] = (int16_t) (s[i] >> 16);
|
||||
}
|
||||
gst_buffer_unmap(buf, &map_info);
|
||||
buf16 = gst_buffer_new_wrapped(d, buf16_size);
|
||||
GST_BUFFER_DURATION(buf16) = GST_FRAMES_TO_CLOCK_TIME(samples * sizeof(int16_t) * channels, rate);
|
||||
buf = buf16;
|
||||
}
|
||||
|
||||
QList<GstBufferConsumer*> consumers;
|
||||
{
|
||||
QMutexLocker l(&instance->buffer_consumers_mutex_);
|
||||
consumers = instance->buffer_consumers_;
|
||||
}
|
||||
|
||||
for (GstBufferConsumer *consumer : consumers) {
|
||||
gst_buffer_ref(buf);
|
||||
consumer->ConsumeBuffer(buf, instance->id(), format);
|
||||
}
|
||||
|
||||
if (buf16) {
|
||||
gst_buffer_unref(buf16);
|
||||
}
|
||||
|
||||
// Calculate the end time of this buffer so we can stop playback if it's after the end time of this song.
|
||||
if (instance->end_offset_nanosec_ > 0) {
|
||||
quint64 start_time = GST_BUFFER_TIMESTAMP(buf) - instance->segment_start_;
|
||||
quint64 duration = GST_BUFFER_DURATION(buf);
|
||||
quint64 end_time = start_time + duration;
|
||||
|
||||
if (end_time > instance->end_offset_nanosec_) {
|
||||
if (instance->has_next_valid_url() && instance->next_stream_url_ == instance->stream_url_ && instance->next_beginning_offset_nanosec_ == instance->end_offset_nanosec_) {
|
||||
// The "next" song is actually the next segment of this file - so cheat and keep on playing, but just tell the Engine we've moved on.
|
||||
instance->end_offset_nanosec_ = instance->next_end_offset_nanosec_;
|
||||
instance->next_stream_url_.clear();
|
||||
instance->next_original_url_.clear();
|
||||
instance->next_beginning_offset_nanosec_ = 0;
|
||||
instance->next_end_offset_nanosec_ = 0;
|
||||
|
||||
// GstEngine will try to seek to the start of the new section, but we're already there so ignore it.
|
||||
instance->ignore_next_seek_ = true;
|
||||
emit instance->EndOfStreamReached(instance->id(), true);
|
||||
}
|
||||
else {
|
||||
// There's no next song
|
||||
emit instance->EndOfStreamReached(instance->id(), false);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
return GST_PAD_PROBE_OK;
|
||||
|
||||
}
|
||||
|
||||
GstPadProbeReturn GstEnginePipeline::EventHandoffCallback(GstPad*, GstPadProbeInfo *info, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
|
||||
GstEvent *e = gst_pad_probe_info_get_event(info);
|
||||
|
||||
qLog(Debug) << instance->id() << "event" << GST_EVENT_TYPE_NAME(e);
|
||||
|
||||
switch (GST_EVENT_TYPE(e)) {
|
||||
case GST_EVENT_SEGMENT:
|
||||
if (!instance->segment_start_received_) {
|
||||
// The segment start time is used to calculate the proper offset of data buffers from the start of the stream
|
||||
const GstSegment *segment = nullptr;
|
||||
gst_event_parse_segment(e, &segment);
|
||||
instance->segment_start_ = segment->start;
|
||||
instance->segment_start_received_ = true;
|
||||
}
|
||||
break;
|
||||
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return GST_PAD_PROBE_OK;
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::AboutToFinishCallback(GstPlayBin*, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
|
||||
if (instance->has_next_valid_url() && !instance->next_uri_set_) {
|
||||
// Set the next uri. When the current song ends it will be played automatically and a STREAM_START message is send to the bus.
|
||||
// When the next uri is not playable an error message is send when the pipeline goes to PLAY (or PAUSE) state or immediately if it is currently in PLAY state.
|
||||
instance->next_uri_set_ = true;
|
||||
g_object_set(G_OBJECT(instance->pipeline_), "uri", instance->next_stream_url_.constData(), nullptr);
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::SourceSetupCallback(GstPlayBin *bin, GParamSpec *, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
|
||||
GstElement *element;
|
||||
g_object_get(bin, "source", &element, nullptr);
|
||||
if (!element) {
|
||||
return;
|
||||
}
|
||||
|
||||
if (g_object_class_find_property(G_OBJECT_GET_CLASS(element), "device") && !instance->source_device().isEmpty()) {
|
||||
// Gstreamer is not able to handle device in URL (referring to Gstreamer documentation, this might be added in the future).
|
||||
// Despite that, for now we include device inside URL: we decompose it during Init and set device here, when this callback is called.
|
||||
g_object_set(element, "device", instance->source_device().toLocal8Bit().constData(), nullptr);
|
||||
}
|
||||
|
||||
if (g_object_class_find_property(G_OBJECT_GET_CLASS(element), "user-agent")) {
|
||||
QString user_agent = QString("%1 %2").arg(QCoreApplication::applicationName(), QCoreApplication::applicationVersion());
|
||||
g_object_set(element, "user-agent", user_agent.toUtf8().constData(), nullptr);
|
||||
g_object_set(element, "ssl-strict", FALSE, nullptr);
|
||||
}
|
||||
|
||||
// If the pipeline was buffering we stop that now.
|
||||
if (instance->buffering_) {
|
||||
instance->buffering_ = false;
|
||||
emit instance->BufferingFinished();
|
||||
instance->SetState(GST_STATE_PLAYING);
|
||||
}
|
||||
|
||||
g_object_unref(element);
|
||||
|
||||
}
|
||||
|
||||
qint64 GstEnginePipeline::position() const {
|
||||
|
||||
if (pipeline_is_initialised_)
|
||||
gst_element_query_position(pipeline_, GST_FORMAT_TIME, &last_known_position_ns_);
|
||||
|
||||
@ -942,6 +951,7 @@ qint64 GstEnginePipeline::position() const {
|
||||
}
|
||||
|
||||
qint64 GstEnginePipeline::length() const {
|
||||
|
||||
gint64 value = 0;
|
||||
gst_element_query_duration(pipeline_, GST_FORMAT_TIME, &value);
|
||||
|
||||
@ -977,8 +987,6 @@ bool GstEnginePipeline::Seek(const qint64 nanosec) {
|
||||
}
|
||||
|
||||
if (next_uri_set_) {
|
||||
qDebug() << "MYTODO: gstenginepipeline.seek: seeking after Transition";
|
||||
|
||||
pending_seek_nanosec_ = nanosec;
|
||||
SetState(GST_STATE_READY);
|
||||
return true;
|
||||
@ -993,7 +1001,6 @@ bool GstEnginePipeline::Seek(const qint64 nanosec) {
|
||||
void GstEnginePipeline::SetVolume(const int percent) {
|
||||
|
||||
if (!volume_) return;
|
||||
|
||||
volume_percent_ = percent;
|
||||
UpdateVolume();
|
||||
|
||||
@ -1015,23 +1022,18 @@ void GstEnginePipeline::UpdateVolume() {
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::SetStereoBalance(const bool enabled, const float value) {
|
||||
void GstEnginePipeline::SetStereoBalance(const float value) {
|
||||
|
||||
stereo_balance_enabled_ = enabled;
|
||||
if (enabled) {
|
||||
stereo_balance_ = value;
|
||||
}
|
||||
else {
|
||||
stereo_balance_ = 0.0f;
|
||||
}
|
||||
UpdateStereoBalance();
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::SetEqualizerEnabled(bool enabled) {
|
||||
void GstEnginePipeline::UpdateStereoBalance() {
|
||||
|
||||
eq_enabled_ = enabled;
|
||||
UpdateEqualizer();
|
||||
if (audiopanorama_) {
|
||||
g_object_set(G_OBJECT(audiopanorama_), "panorama", stereo_balance_, nullptr);
|
||||
}
|
||||
|
||||
}
|
||||
|
||||
@ -1043,12 +1045,6 @@ void GstEnginePipeline::SetEqualizerParams(const int preamp, const QList<int>& b
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::UpdateStereoBalance() {
|
||||
if (audiopanorama_) {
|
||||
g_object_set(G_OBJECT(audiopanorama_), "panorama", stereo_balance_, nullptr);
|
||||
}
|
||||
}
|
||||
|
||||
void GstEnginePipeline::UpdateEqualizer() {
|
||||
|
||||
if (!equalizer_ || !equalizer_preamp_) return;
|
||||
@ -1171,7 +1167,6 @@ void GstEnginePipeline::SetNextUrl(const QByteArray &stream_url, const QUrl &ori
|
||||
void GstEnginePipeline::StreamDiscovered(GstDiscoverer*, GstDiscovererInfo *info, GError*, gpointer self) {
|
||||
|
||||
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
|
||||
if (!instance) return;
|
||||
|
||||
QString discovered_url(gst_discoverer_info_get_uri(info));
|
||||
|
||||
|
@ -69,7 +69,9 @@ class GstEnginePipeline : public QObject {
|
||||
|
||||
// Call these setters before Init
|
||||
void set_output_device(const QString &sink, const QVariant &device);
|
||||
void set_volume_control(const bool volume_control);
|
||||
void set_volume_enabled(const bool enabled);
|
||||
void set_stereo_balancer_enabled(const bool enabled);
|
||||
void set_equalizer_enabled(const bool enabled);
|
||||
void set_replaygain(const bool enabled, const int mode, const float preamp, const bool compression);
|
||||
void set_buffer_duration_nanosec(qint64 duration_nanosec);
|
||||
void set_buffer_min_fill(int percent);
|
||||
@ -85,10 +87,10 @@ class GstEnginePipeline : public QObject {
|
||||
// Control the music playback
|
||||
QFuture<GstStateChangeReturn> SetState(const GstState state);
|
||||
Q_INVOKABLE bool Seek(const qint64 nanosec);
|
||||
void SetEqualizerEnabled(const bool enabled);
|
||||
void SetEqualizerParams(const int preamp, const QList<int> &band_gains);
|
||||
void SetVolume(const int percent);
|
||||
void SetStereoBalance(const bool enabled, const float value);
|
||||
void SetStereoBalance(const float value);
|
||||
void SetEqualizerParams(const int preamp, const QList<int> &band_gains);
|
||||
|
||||
void StartFader(const qint64 duration_nanosec, const QTimeLine::Direction direction = QTimeLine::Forward, const QTimeLine::CurveShape shape = QTimeLine::LinearCurve, const bool use_fudge_timer = true);
|
||||
|
||||
// If this is set then it will be loaded automatically when playback finishes for gapless playback
|
||||
@ -101,6 +103,7 @@ class GstEnginePipeline : public QObject {
|
||||
QByteArray stream_url() const { return stream_url_; }
|
||||
QUrl original_url() const { return original_url_; }
|
||||
bool is_valid() const { return valid_; }
|
||||
|
||||
// Please note that this method (unlike GstEngine's.position()) is multiple-section media unaware.
|
||||
qint64 position() const;
|
||||
// Please note that this method (unlike GstEngine's.length()) is multiple-section media unaware.
|
||||
@ -135,17 +138,21 @@ signals:
|
||||
void timerEvent(QTimerEvent*);
|
||||
|
||||
private:
|
||||
bool InitAudioBin();
|
||||
|
||||
// Static callbacks. The GstEnginePipeline instance is passed in the last argument.
|
||||
static GstPadProbeReturn EventHandoffCallback(GstPad*, GstPadProbeInfo*, gpointer);
|
||||
static void SourceSetupCallback(GstPlayBin*, GParamSpec* pspec, gpointer);
|
||||
static void NewPadCallback(GstElement*, GstPad*, gpointer);
|
||||
static GstPadProbeReturn PlaybinProbe(GstPad*, GstPadProbeInfo*, gpointer);
|
||||
static GstPadProbeReturn HandoffCallback(GstPad*, GstPadProbeInfo*, gpointer);
|
||||
static void AboutToFinishCallback(GstPlayBin*, gpointer);
|
||||
static GstBusSyncReply BusCallbackSync(GstBus*, GstMessage*, gpointer);
|
||||
static gboolean BusCallback(GstBus*, GstMessage*, gpointer);
|
||||
static void NewPadCallback(GstElement*, GstPad*, gpointer);
|
||||
static GstPadProbeReturn HandoffCallback(GstPad*, GstPadProbeInfo*, gpointer);
|
||||
static GstPadProbeReturn SourceHandoffCallback(GstPad*, GstPadProbeInfo*, gpointer);
|
||||
static GstPadProbeReturn EventHandoffCallback(GstPad*, GstPadProbeInfo*, gpointer);
|
||||
static void AboutToFinishCallback(GstPlayBin*, gpointer);
|
||||
static GstPadProbeReturn PlaybinProbe(GstPad*, GstPadProbeInfo*, gpointer);
|
||||
static void SourceSetupCallback(GstPlayBin*, GParamSpec* pspec, gpointer);
|
||||
static void TaskEnterCallback(GstTask*, GThread*, gpointer);
|
||||
static void StreamDiscovered(GstDiscoverer *discoverer, GstDiscovererInfo *info, GError *err, gpointer instance);
|
||||
static void StreamDiscoveryFinished(GstDiscoverer *discoverer, gpointer instance);
|
||||
static QString GSTdiscovererErrorMessage(GstDiscovererResult result);
|
||||
|
||||
void TagMessageReceived(GstMessage*);
|
||||
void ErrorMessageReceived(GstMessage*);
|
||||
@ -158,15 +165,9 @@ signals:
|
||||
QString ParseStrTag(GstTagList *list, const char *tag) const;
|
||||
guint ParseUIntTag(GstTagList *list, const char *tag) const;
|
||||
|
||||
bool InitAudioBin();
|
||||
|
||||
void UpdateVolume();
|
||||
void UpdateEqualizer();
|
||||
void UpdateStereoBalance();
|
||||
|
||||
static void StreamDiscovered(GstDiscoverer *discoverer, GstDiscovererInfo *info, GError *err, gpointer instance);
|
||||
static void StreamDiscoveryFinished(GstDiscoverer *discoverer, gpointer instance);
|
||||
static QString GSTdiscovererErrorMessage(GstDiscovererResult result);
|
||||
void UpdateEqualizer();
|
||||
|
||||
private slots:
|
||||
void FaderTimelineFinished();
|
||||
@ -191,21 +192,21 @@ signals:
|
||||
bool valid_;
|
||||
QString output_;
|
||||
QVariant device_;
|
||||
bool volume_control_;
|
||||
bool volume_enabled_;
|
||||
bool stereo_balancer_enabled_;
|
||||
bool eq_enabled_;
|
||||
bool rg_enabled_;
|
||||
|
||||
// Stereo balance.
|
||||
// Stereo balance:
|
||||
// From -1.0 - 1.0
|
||||
// -1.0 is left, 1.0 is right.
|
||||
bool stereo_balance_enabled_;
|
||||
float stereo_balance_;
|
||||
|
||||
// Equalizer
|
||||
bool eq_enabled_;
|
||||
int eq_preamp_;
|
||||
QList<int> eq_band_gains_;
|
||||
|
||||
// ReplayGain
|
||||
bool rg_enabled_;
|
||||
int rg_mode_;
|
||||
float rg_preamp_;
|
||||
bool rg_compression_;
|
||||
@ -276,9 +277,9 @@ signals:
|
||||
GstElement *equalizer_preamp_;
|
||||
GstDiscoverer *discoverer_;
|
||||
|
||||
int about_to_finish_cb_id_;
|
||||
int pad_added_cb_id_;
|
||||
int notify_source_cb_id_;
|
||||
int about_to_finish_cb_id_;
|
||||
int bus_cb_id_;
|
||||
int discovery_finished_cb_id_;
|
||||
int discovery_discovered_cb_id_;
|
||||
|
@ -74,16 +74,14 @@ Equalizer::Equalizer(QWidget *parent)
|
||||
// Must be done before the signals are connected
|
||||
ReloadSettings();
|
||||
|
||||
connect(ui_->enable, SIGNAL(toggled(bool)), SIGNAL(EnabledChanged(bool)));
|
||||
connect(ui_->enable, SIGNAL(toggled(bool)), ui_->slider_container, SLOT(setEnabled(bool)));
|
||||
connect(ui_->enable, SIGNAL(toggled(bool)), SLOT(Save()));
|
||||
connect(ui_->enable_equalizer, SIGNAL(toggled(bool)), SLOT(EqualizerEnabledChangedSlot(bool)));
|
||||
|
||||
connect(ui_->preset, SIGNAL(currentIndexChanged(int)), SLOT(PresetChanged(int)));
|
||||
connect(ui_->preset_save, SIGNAL(clicked()), SLOT(SavePreset()));
|
||||
connect(ui_->preset_del, SIGNAL(clicked()), SLOT(DelPreset()));
|
||||
|
||||
connect(ui_->enable_stereo_balancer, SIGNAL(toggled(bool)), SLOT(Save()));
|
||||
connect(ui_->enable_stereo_balancer, SIGNAL(toggled(bool)), ui_->balance_slider, SLOT(setEnabled(bool)));
|
||||
connect(ui_->balance_slider, SIGNAL(valueChanged(int)), SLOT(StereoSliderChanged(int)));
|
||||
connect(ui_->enable_stereo_balancer, SIGNAL(toggled(bool)), SLOT(StereoBalancerEnabledChangedSlot(bool)));
|
||||
connect(ui_->stereo_balance_slider, SIGNAL(valueChanged(int)), SLOT(StereoBalanceSliderChanged(int)));
|
||||
|
||||
QShortcut *close = new QShortcut(QKeySequence::Close, this);
|
||||
connect(close, SIGNAL(activated()), SLOT(close()));
|
||||
@ -119,16 +117,16 @@ void Equalizer::ReloadSettings() {
|
||||
if (selected_index != -1) ui_->preset->setCurrentIndex(selected_index);
|
||||
|
||||
// Enabled?
|
||||
ui_->enable->setChecked(s.value("enabled", false).toBool());
|
||||
ui_->slider_container->setEnabled(ui_->enable->isChecked());
|
||||
ui_->enable_equalizer->setChecked(s.value("enabled", false).toBool());
|
||||
ui_->slider_container->setEnabled(ui_->enable_equalizer->isChecked());
|
||||
|
||||
ui_->enable_stereo_balancer->setChecked(s.value("enable_stereo_balancer", false).toBool());
|
||||
ui_->slider_label_layout->setEnabled(ui_->enable_stereo_balancer->isChecked());
|
||||
ui_->balance_slider->setEnabled(ui_->enable_stereo_balancer->isChecked());
|
||||
ui_->stereo_balance_slider->setEnabled(ui_->enable_stereo_balancer->isChecked());
|
||||
|
||||
int stereo_balance = s.value("stereo_balance", 0).toInt();
|
||||
ui_->balance_slider->setValue(stereo_balance);
|
||||
StereoSliderChanged(stereo_balance);
|
||||
ui_->stereo_balance_slider->setValue(stereo_balance);
|
||||
StereoBalanceSliderChanged(stereo_balance);
|
||||
|
||||
PresetChanged(selected_preset);
|
||||
|
||||
@ -191,7 +189,7 @@ void Equalizer::PresetChanged(const QString& name) {
|
||||
for (int i = 0; i < kBands; ++i) gain_[i]->set_value(p.gain[i]);
|
||||
loading_ = false;
|
||||
|
||||
ParametersChanged();
|
||||
EqualizerParametersChangedSlot();
|
||||
Save();
|
||||
|
||||
}
|
||||
@ -241,7 +239,7 @@ EqualizerSlider *Equalizer::AddSlider(const QString &label) {
|
||||
|
||||
EqualizerSlider *ret = new EqualizerSlider(label, ui_->slider_container);
|
||||
ui_->slider_container->layout()->addWidget(ret);
|
||||
connect(ret, SIGNAL(ValueChanged(int)), SLOT(ParametersChanged()));
|
||||
connect(ret, SIGNAL(ValueChanged(int)), SLOT(EqualizerParametersChangedSlot()));
|
||||
|
||||
return ret;
|
||||
|
||||
@ -252,7 +250,7 @@ bool Equalizer::is_stereo_balancer_enabled() const {
|
||||
}
|
||||
|
||||
bool Equalizer::is_equalizer_enabled() const {
|
||||
return ui_->enable->isChecked();
|
||||
return ui_->enable_equalizer->isChecked();
|
||||
}
|
||||
|
||||
int Equalizer::preamp_value() const {
|
||||
@ -279,13 +277,41 @@ Equalizer::Params Equalizer::current_params() const {
|
||||
}
|
||||
|
||||
float Equalizer::stereo_balance() const {
|
||||
return qBound(-1.0f, ui_->balance_slider->value() / 100.0f, 1.0f);
|
||||
return qBound(-1.0f, ui_->stereo_balance_slider->value() / 100.0f, 1.0f);
|
||||
}
|
||||
|
||||
void Equalizer::ParametersChanged() {
|
||||
if (loading_) return;
|
||||
void Equalizer::StereoBalancerEnabledChangedSlot(const bool enabled) {
|
||||
|
||||
if (!enabled) {
|
||||
ui_->stereo_balance_slider->setValue(0);
|
||||
emit StereoBalanceChanged(stereo_balance());
|
||||
}
|
||||
ui_->stereo_balance_slider->setEnabled(enabled);
|
||||
emit StereoBalancerEnabledChanged(enabled);
|
||||
Save();
|
||||
|
||||
}
|
||||
|
||||
void Equalizer::StereoBalanceSliderChanged(int) {
|
||||
|
||||
emit StereoBalanceChanged(stereo_balance());
|
||||
Save();
|
||||
|
||||
}
|
||||
|
||||
void Equalizer::EqualizerEnabledChangedSlot(const bool enabled) {
|
||||
|
||||
emit EqualizerEnabledChanged(enabled);
|
||||
ui_->slider_container->setEnabled(enabled);
|
||||
Save();
|
||||
|
||||
}
|
||||
|
||||
void Equalizer::EqualizerParametersChangedSlot() {
|
||||
|
||||
if (loading_) return;
|
||||
emit EqualizerParametersChanged(preamp_value(), gain_values());
|
||||
|
||||
emit ParametersChanged(preamp_value(), gain_values());
|
||||
}
|
||||
|
||||
void Equalizer::Save() {
|
||||
@ -307,16 +333,14 @@ void Equalizer::Save() {
|
||||
s.setValue("selected_preset", ui_->preset->itemData(ui_->preset->currentIndex()).toString());
|
||||
|
||||
// Enabled?
|
||||
s.setValue("enabled", ui_->enable->isChecked());
|
||||
s.setValue("enabled", ui_->enable_equalizer->isChecked());
|
||||
|
||||
s.setValue("enable_stereo_balancer", ui_->enable_stereo_balancer->isChecked());
|
||||
s.setValue("stereo_balance", ui_->balance_slider->value());
|
||||
s.setValue("stereo_balance", ui_->stereo_balance_slider->value());
|
||||
|
||||
}
|
||||
|
||||
void Equalizer::closeEvent(QCloseEvent *e) {
|
||||
|
||||
Q_UNUSED(e);
|
||||
void Equalizer::closeEvent(QCloseEvent*) {
|
||||
|
||||
QString name = ui_->preset->currentText();
|
||||
if (!presets_.contains(name)) return;
|
||||
@ -331,8 +355,7 @@ Equalizer::Params::Params() : preamp(0) {
|
||||
for (int i = 0; i < Equalizer::kBands; ++i) gain[i] = 0;
|
||||
}
|
||||
|
||||
Equalizer::Params::Params(int g0, int g1, int g2, int g3, int g4, int g5, int g6, int g7, int g8, int g9, int pre)
|
||||
: preamp(pre) {
|
||||
Equalizer::Params::Params(int g0, int g1, int g2, int g3, int g4, int g5, int g6, int g7, int g8, int g9, int pre) : preamp(pre) {
|
||||
gain[0] = g0;
|
||||
gain[1] = g1;
|
||||
gain[2] = g2;
|
||||
@ -357,12 +380,6 @@ bool Equalizer::Params::operator !=(const Equalizer::Params& other) const {
|
||||
return ! (*this == other);
|
||||
}
|
||||
|
||||
void Equalizer::StereoSliderChanged(int value) {
|
||||
Q_UNUSED(value);
|
||||
emit StereoBalanceChanged(is_stereo_balancer_enabled(), stereo_balance());
|
||||
Save();
|
||||
}
|
||||
|
||||
QDataStream &operator<<(QDataStream& s, const Equalizer::Params& p) {
|
||||
s << p.preamp;
|
||||
for (int i = 0; i < Equalizer::kBands; ++i) s << p.gain[i];
|
||||
|
@ -69,21 +69,24 @@ class Equalizer : public QDialog {
|
||||
float stereo_balance() const;
|
||||
|
||||
signals:
|
||||
void EnabledChanged(bool enabled);
|
||||
void ParametersChanged(int preamp, const QList<int> &band_gains);
|
||||
void StereoBalanceChanged(bool enabled, float balance);
|
||||
void StereoBalancerEnabledChanged(const bool enabled);
|
||||
void StereoBalanceChanged(const float balance);
|
||||
void EqualizerEnabledChanged(const bool enabled);
|
||||
void EqualizerParametersChanged(const int preamp, const QList<int> &band_gains);
|
||||
|
||||
protected:
|
||||
void closeEvent(QCloseEvent*);
|
||||
|
||||
private slots:
|
||||
void ParametersChanged();
|
||||
void StereoBalancerEnabledChangedSlot(const bool enabled);
|
||||
void StereoBalanceSliderChanged(const int value);
|
||||
void EqualizerEnabledChangedSlot(const bool enabled);
|
||||
void EqualizerParametersChangedSlot();
|
||||
void PresetChanged(const QString &name);
|
||||
void PresetChanged(int index);
|
||||
void SavePreset();
|
||||
void DelPreset();
|
||||
void Save();
|
||||
void StereoSliderChanged(int value);
|
||||
|
||||
private:
|
||||
EqualizerSlider *AddSlider(const QString &label);
|
||||
|
@ -61,7 +61,7 @@
|
||||
</widget>
|
||||
</item>
|
||||
<item>
|
||||
<widget class="QCheckBox" name="enable">
|
||||
<widget class="QCheckBox" name="enable_equalizer">
|
||||
<property name="text">
|
||||
<string>Enable equalizer</string>
|
||||
</property>
|
||||
@ -133,7 +133,7 @@
|
||||
</layout>
|
||||
</item>
|
||||
<item>
|
||||
<widget class="QSlider" name="balance_slider">
|
||||
<widget class="QSlider" name="stereo_balance_slider">
|
||||
<property name="minimum">
|
||||
<number>-100</number>
|
||||
</property>
|
||||
|
Loading…
x
Reference in New Issue
Block a user