Refactor gstreamer engine code, equalizer and fix stereo balancer
This commit is contained in:
Jonas Kvinge
@@ -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);
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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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// 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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default:
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break;
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}
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if (!InitAudioBin()) return false;
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return GST_PAD_PROBE_OK;
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// Set playbin's sink to be our costum 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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}
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return true;
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void GstEnginePipeline::SourceSetupCallback(GstPlayBin *bin, GParamSpec *, gpointer self) {
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GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
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GstElement *element = nullptr;
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g_object_get(bin, "source", &element, nullptr);
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if (!element) {
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return;
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}
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if (g_object_class_find_property(G_OBJECT_GET_CLASS(element), "device") && !instance->source_device().isEmpty()) {
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// Gstreamer is not able to handle device in URL (referring to Gstreamer documentation, this might be added in the future).
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// Despite that, for now we include device inside URL: we decompose it during Init and set device here, when this callback is called.
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g_object_set(element, "device", instance->source_device().toLocal8Bit().constData(), nullptr);
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}
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if (g_object_class_find_property(G_OBJECT_GET_CLASS(element), "user-agent")) {
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QString user_agent = QString("%1 %2").arg(QCoreApplication::applicationName(), QCoreApplication::applicationVersion());
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g_object_set(element, "user-agent", user_agent.toUtf8().constData(), nullptr);
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g_object_set(element, "ssl-strict", FALSE, nullptr);
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}
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// If the pipeline was buffering we stop that now.
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if (instance->buffering_) {
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instance->buffering_ = false;
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emit instance->BufferingFinished();
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instance->SetState(GST_STATE_PLAYING);
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}
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g_object_unref(element);
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}
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void GstEnginePipeline::NewPadCallback(GstElement*, GstPad *pad, gpointer self) {
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GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
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GstPad *const audiopad = gst_element_get_static_pad(instance->audiobin_, "sink");
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// Link playbin's sink pad to audiobin's src pad.
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if (GST_PAD_IS_LINKED(audiopad)) {
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qLog(Warning) << instance->id() << "audiopad is already linked, unlinking old pad";
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gst_pad_unlink(audiopad, GST_PAD_PEER(audiopad));
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}
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gst_pad_link(pad, audiopad);
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gst_object_unref(audiopad);
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// 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.
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// "Running time" is the time since the last flushing seek.
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GstClockTime running_time = gst_segment_to_running_time(&instance->last_playbin_segment_, GST_FORMAT_TIME, instance->last_playbin_segment_.position);
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gst_pad_set_offset(pad, running_time);
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// Add a probe to the pad so we can update last_playbin_segment_.
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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);
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instance->pipeline_is_connected_ = true;
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if (instance->pending_seek_nanosec_ != -1 && instance->pipeline_is_initialised_) {
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QMetaObject::invokeMethod(instance, "Seek", Qt::QueuedConnection, Q_ARG(qint64, instance->pending_seek_nanosec_));
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}
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}
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GstPadProbeReturn GstEnginePipeline::PlaybinProbe(GstPad *pad, GstPadProbeInfo *info, gpointer data) {
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GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(data);
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const GstPadProbeType info_type = GST_PAD_PROBE_INFO_TYPE(info);
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if (info_type & GST_PAD_PROBE_TYPE_BUFFER) {
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// 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.
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GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER(info);
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GstClockTime timestamp = GST_BUFFER_TIMESTAMP(buffer);
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GstClockTime duration = GST_BUFFER_DURATION(buffer);
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if (timestamp == GST_CLOCK_TIME_NONE) {
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timestamp = instance->last_playbin_segment_.position;
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}
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if (duration != GST_CLOCK_TIME_NONE) {
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timestamp += duration;
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}
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instance->last_playbin_segment_.position = timestamp;
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}
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else if (info_type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) {
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GstEvent *event = GST_PAD_PROBE_INFO_EVENT(info);
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GstEventType event_type = GST_EVENT_TYPE(event);
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if (event_type == GST_EVENT_SEGMENT) {
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// A new segment started, we need to save this to calculate running time offsets later.
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gst_event_copy_segment(event, &instance->last_playbin_segment_);
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}
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else if (event_type == GST_EVENT_FLUSH_START) {
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// A flushing seek resets the running time to 0, so remove any offset we set on this pad before.
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gst_pad_set_offset(pad, 0);
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}
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}
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return GST_PAD_PROBE_OK;
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}
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GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad *pad, GstPadProbeInfo *info, gpointer self) {
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GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
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GstCaps *caps = gst_pad_get_current_caps(pad);
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GstStructure *structure = gst_caps_get_structure(caps, 0);
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QString format = QString(gst_structure_get_string(structure, "format"));
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int channels = 0;
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int rate = 0;
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gst_structure_get_int(structure, "channels", &channels);
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gst_structure_get_int(structure, "rate", &rate);
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GstBuffer *buf = gst_pad_probe_info_get_buffer(info);
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GstBuffer *buf16 = nullptr;
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if (format.startsWith("S32")) {
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GstMapInfo map_info;
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gst_buffer_map(buf, &map_info, GST_MAP_READ);
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int32_t *s = (int32_t*) map_info.data;
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int samples = (map_info.size / sizeof(int32_t)) / channels;
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int buf16_size = samples * sizeof(int16_t) * channels;
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int16_t *d = (int16_t*) g_malloc(buf16_size);
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memset(d, 0, buf16_size);
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for (int i = 0 ; i < (samples * 2) ; ++i) {
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d[i] = (int16_t) (s[i] >> 16);
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}
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gst_buffer_unmap(buf, &map_info);
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buf16 = gst_buffer_new_wrapped(d, buf16_size);
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GST_BUFFER_DURATION(buf16) = GST_FRAMES_TO_CLOCK_TIME(samples * sizeof(int16_t) * channels, rate);
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buf = buf16;
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}
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QList<GstBufferConsumer*> consumers;
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{
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QMutexLocker l(&instance->buffer_consumers_mutex_);
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consumers = instance->buffer_consumers_;
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}
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for (GstBufferConsumer *consumer : consumers) {
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gst_buffer_ref(buf);
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||||
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;
|
||||
}
|
||||
stereo_balance_ = value;
|
||||
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;
|
||||
@@ -1168,10 +1164,9 @@ void GstEnginePipeline::SetNextUrl(const QByteArray &stream_url, const QUrl &ori
|
||||
|
||||
}
|
||||
|
||||
void GstEnginePipeline::StreamDiscovered(GstDiscoverer *, GstDiscovererInfo *info, GError *, gpointer self) {
|
||||
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));
|
||||
|
||||
|
||||
Reference in New Issue
Block a user