/* * Strawberry Music Player * This file was part of Clementine. * Copyright 2010, David Sansome * Copyright 2018-2021, Jonas Kvinge * * Strawberry is free software: you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation, either version 3 of the License, or * (at your option) any later version. * * Strawberry is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with Strawberry. If not, see . * */ #ifndef GSTENGINEPIPELINE_H #define GSTENGINEPIPELINE_H #include "config.h" #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include class QTimerEvent; class GstBufferConsumer; namespace Engine { struct SimpleMetaBundle; } // namespace Engine struct GstPlayBin; class GstEnginePipeline : public QObject { Q_OBJECT public: explicit GstEnginePipeline(QObject *parent = nullptr); ~GstEnginePipeline() override; // Globally unique across all pipelines. int id() const { return id_; } // Call these setters before Init void set_output_device(const QString &output, const QVariant &device); 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 double preamp, const double fallbackgain, const bool compression); void set_buffer_duration_nanosec(const quint64 duration_nanosec); void set_buffer_low_watermark(const double value); void set_buffer_high_watermark(const double value); void set_proxy_settings(const QString &address, const bool authentication, const QString &user, const QString &pass); void set_channels(const bool enabled, const int channels); void set_bs2b_enabled(const bool enabled); void set_strict_ssl_enabled(const bool enabled); void set_fading_enabled(const bool enabled); // Creates the pipeline, returns false on error bool InitFromUrl(const QByteArray &stream_url, const QUrl &original_url, const qint64 end_nanosec, QString &error); // GstBufferConsumers get fed audio data. Thread-safe. void AddBufferConsumer(GstBufferConsumer *consumer); void RemoveBufferConsumer(GstBufferConsumer *consumer); void RemoveAllBufferConsumers(); // Control the music playback QFuture SetState(const GstState state); Q_INVOKABLE bool Seek(const qint64 nanosec); void SetVolume(const uint volume_percent); void SetStereoBalance(const float value); void SetEqualizerParams(const int preamp, const QList &band_gains); void StartFader(const qint64 duration_nanosec, const QTimeLine::Direction direction = QTimeLine::Forward, const QEasingCurve::Type shape = QEasingCurve::Linear, const bool use_fudge_timer = true); // If this is set then it will be loaded automatically when playback finishes for gapless playback void SetNextUrl(const QByteArray &stream_url, const QUrl &original_url, qint64 beginning_nanosec, qint64 end_nanosec); bool has_next_valid_url() const { return !next_stream_url_.isEmpty(); } void SetSourceDevice(const QString &device) { source_device_ = device; } // Get information about the music playback QByteArray stream_url() const { return stream_url_; } QByteArray next_stream_url() const { return next_stream_url_; } QUrl original_url() const { return original_url_; } QUrl next_original_url() const { return next_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. qint64 length() const; // Returns this pipeline's state. May return GST_STATE_NULL if the state check timed out. The timeout value is a reasonable default. GstState state() const; qint64 segment_start() const { return segment_start_; } // Don't allow the user to change the playback state (playing/paused) while the pipeline is buffering. bool is_buffering() const { return buffering_; } QByteArray redirect_url() const { return redirect_url_; } QString source_device() const { return source_device_; } public slots: void SetFaderVolume(const qreal volume); signals: void Error(int pipeline_id, int domain, int error_code, QString message, QString debug); void EndOfStreamReached(int pipeline_id, bool has_next_track); void MetadataFound(int pipeline_id, const Engine::SimpleMetaBundle &bundle); void VolumeChanged(uint volume); void FaderFinished(); void BufferingStarted(); void BufferingProgress(int percent); void BufferingFinished(); protected: void timerEvent(QTimerEvent*) override; private: GstElement *CreateElement(const QString &factory_name, const QString &name, GstElement *bin, QString &error) const; bool InitAudioBin(QString &error); void SetupVolume(GstElement *element); // Static callbacks. The GstEnginePipeline instance is passed in the last argument. static GstPadProbeReturn UpstreamEventsProbeCallback(GstPad *pad, GstPadProbeInfo *info, gpointer self); static GstPadProbeReturn BufferProbeCallback(GstPad *pad, GstPadProbeInfo *info, gpointer self); static GstPadProbeReturn PlaybinProbeCallback(GstPad *pad, GstPadProbeInfo *info, gpointer self); static void ElementAddedCallback(GstBin *bin, GstBin*, GstElement *element, gpointer self); static void PadAddedCallback(GstElement *element, GstPad *pad, gpointer self); static void SourceSetupCallback(GstElement *playbin, GstElement *source, gpointer self); static void NotifyVolumeCallback(GstElement *element, GParamSpec *param_spec, gpointer self); static void AboutToFinishCallback(GstPlayBin *playbin, gpointer self); static GstBusSyncReply BusSyncCallback(GstBus *bus, GstMessage *msg, gpointer self); static gboolean BusWatchCallback(GstBus *bus, GstMessage *msg, gpointer self); static void TaskEnterCallback(GstTask *task, GThread *thread, gpointer self); void TagMessageReceived(GstMessage *msg); void ErrorMessageReceived(GstMessage *msg); void ElementMessageReceived(GstMessage *msg); void StateChangedMessageReceived(GstMessage *msg); void BufferingMessageReceived(GstMessage *msg); void StreamStatusMessageReceived(GstMessage *msg); void StreamStartMessageReceived(); static QString ParseStrTag(GstTagList *list, const char *tag); static guint ParseUIntTag(GstTagList *list, const char *tag); void UpdateStereoBalance(); void UpdateEqualizer(); private slots: void FaderTimelineFinished(); private: static const int kGstStateTimeoutNanosecs; static const int kFaderFudgeMsec; static const int kEqBandCount; static const int kEqBandFrequencies[]; // Using == to compare two pipelines is a bad idea, because new ones often get created in the same address as old ones. This ID will be unique for each pipeline. // Threading warning: access to the static ID field isn't protected by a mutex because all pipeline creation is currently done in the main thread. static int sId; int id_; // General settings for the pipeline bool valid_; QString output_; QVariant device_; bool volume_enabled_; bool stereo_balancer_enabled_; bool eq_enabled_; bool rg_enabled_; bool fading_enabled_; // Stereo balance: // From -1.0 - 1.0 // -1.0 is left, 1.0 is right. float stereo_balance_; // Equalizer int eq_preamp_; QList eq_band_gains_; // ReplayGain int rg_mode_; double rg_preamp_; double rg_fallbackgain_; bool rg_compression_; // Buffering quint64 buffer_duration_nanosec_; double buffer_low_watermark_; double buffer_high_watermark_; bool buffering_; // Proxy QString proxy_address_; bool proxy_authentication_; QString proxy_user_; QString proxy_pass_; // Channels bool channels_enabled_; int channels_; // Options bool bs2b_enabled_; bool strict_ssl_enabled_; // These get called when there is a new audio buffer available QList buffer_consumers_; QMutex buffer_consumers_mutex_; qint64 segment_start_; bool segment_start_received_; // The URL that is currently playing, and the URL that is to be preloaded when the current track is close to finishing. QByteArray stream_url_; QUrl original_url_; QByteArray next_stream_url_; QUrl next_original_url_; // If this is > 0 then the pipeline will be forced to stop when playback goes past this position. qint64 end_offset_nanosec_; // We store the beginning and end for the preloading song too, so we can just carry on without reloading the file if the sections carry on from each other. qint64 next_beginning_offset_nanosec_; qint64 next_end_offset_nanosec_; // Set temporarily when moving to the next contiguous section in a multipart file. bool ignore_next_seek_; // Set temporarily when switching out the decode bin, so metadata doesn't get sent while the Player still thinks it's playing the last song bool ignore_tags_; // When the gstreamer source requests a redirect we store the URL here and callers can pick it up after the state change to PLAYING fails. QByteArray redirect_url_; // When we need to specify the device to use as source (for CD device) QString source_device_; // Seeking while the pipeline is in the READY state doesn't work, so we have to wait until it goes to PAUSED or PLAYING. // Also, we have to wait for the playbin to be connected. bool pipeline_is_initialized_; bool pipeline_is_connected_; qint64 pending_seek_nanosec_; // We can only use gst_element_query_position() when the pipeline is in // PAUSED nor PLAYING state. Whenever we get a new position (e.g. after a correct call to gst_element_query_position() or after a seek), we store // it here so that we can use it when using gst_element_query_position() is not possible. mutable gint64 last_known_position_ns_; // Complete the transition to the next song when it starts playing bool next_uri_set_; gdouble volume_internal_; uint volume_percent_; std::shared_ptr fader_; QBasicTimer fader_fudge_timer_; bool use_fudge_timer_; GstElement *pipeline_; GstElement *audiobin_; GstElement *audiosink_; GstElement *audioqueue_; GstElement *audioqueueconverter_; GstElement *volume_; GstElement *volume_sw_; GstElement *volume_fading_; GstElement *audiopanorama_; GstElement *equalizer_; GstElement *equalizer_preamp_; GstElement *eventprobe_; gulong upstream_events_probe_cb_id_; gulong buffer_probe_cb_id_; gulong playbin_probe_cb_id_; glong element_added_cb_id_; glong pad_added_cb_id_; glong notify_source_cb_id_; glong about_to_finish_cb_id_; glong notify_volume_cb_id_; QThreadPool set_state_threadpool_; GstSegment last_playbin_segment_{}; bool logged_unsupported_analyzer_format_; }; #endif // GSTENGINEPIPELINE_H