Clementine-audio-player-Mac.../src/engines/gstenginepipeline.h

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9.9 KiB
C++

/* This file is part of Clementine.
Copyright 2010, David Sansome <me@davidsansome.com>
Clementine 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.
Clementine 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 Clementine. If not, see <http://www.gnu.org/licenses/>.
*/
#ifndef GSTENGINEPIPELINE_H
#define GSTENGINEPIPELINE_H
#include <QBasicTimer>
#include <QFuture>
#include <QMutex>
#include <QThreadPool>
#include <QTimeLine>
#include <QUrl>
#include <memory>
#include "engine_fwd.h"
#include "gstpipelinebase.h"
#include "playbackrequest.h"
class GstElementDeleter;
class GstEngine;
class BufferConsumer;
struct GstQueue;
struct GstURIDecodeBin;
class GstEnginePipeline : public GstPipelineBase {
Q_OBJECT
public:
GstEnginePipeline(GstEngine* engine);
~GstEnginePipeline();
// Call these setters before Init
void set_output_device(const QString& sink, const QVariant& device);
void set_replaygain(bool enabled, int mode, float preamp, bool compression);
void set_buffer_duration_nanosec(qint64 duration_nanosec);
void set_buffer_min_fill(int percent);
void set_mono_playback(bool enabled);
void set_sample_rate(int rate);
void set_format(const QString& format) { format_ = format; }
// Creates the pipeline, returns false on error
bool InitFromReq(const MediaPlaybackRequest& req, qint64 end_nanosec);
bool InitFromString(const QString& pipeline);
// BufferConsumers get fed audio data. Thread-safe.
void AddBufferConsumer(BufferConsumer* consumer);
void RemoveBufferConsumer(BufferConsumer* consumer);
void RemoveAllBufferConsumers();
// Control the music playback
QFuture<GstStateChangeReturn> SetState(GstState state);
Q_INVOKABLE bool Seek(qint64 nanosec);
void SetEqualizerEnabled(bool enabled);
void SetEqualizerParams(int preamp, const QList<int>& band_gains);
void SetVolume(int percent);
void SetStereoBalance(float value);
void StartFader(qint64 duration_nanosec,
QTimeLine::Direction direction = QTimeLine::Forward,
QTimeLine::CurveShape shape = QTimeLine::LinearCurve,
bool use_fudge_timer = true);
// If this is set then it will be loaded automatically when playback finishes
// for gapless playback
void SetNextReq(const MediaPlaybackRequest& req, qint64 beginning_nanosec,
qint64 end_nanosec);
bool has_next_valid_url() const { return next_.url_.isValid(); }
// Get information about the music playback
QUrl url() const { return current_.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_; }
QUrl redirect_url() const { return redirect_url_; }
QString source_device() const { return source_device_; }
public slots:
void SetVolumeModifier(qreal mod);
signals:
void EndOfStreamReached(int pipeline_id, bool has_next_track);
void MetadataFound(int pipeline_id, const Engine::SimpleMetaBundle& bundle);
// This indicates an error, delegated from GStreamer, in the pipeline.
// The message, domain and error_code are related to GStreamer's GError.
void Error(int pipeline_id, const QString& message, int domain,
int error_code);
void FaderFinished();
void BufferingStarted();
void BufferingProgress(int percent);
void BufferingFinished();
protected:
void timerEvent(QTimerEvent*);
private:
// Static callbacks. The GstEnginePipeline instance is passed in the last
// argument.
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 EventHandoffCallback(GstPad*, GstPadProbeInfo*,
gpointer);
static GstPadProbeReturn DecodebinProbe(GstPad*, GstPadProbeInfo*, gpointer);
static void SourceDrainedCallback(GstURIDecodeBin*, gpointer);
static void SourceSetupCallback(GstURIDecodeBin*, GParamSpec* pspec,
gpointer);
static void TaskEnterCallback(GstTask*, GThread*, gpointer);
static QByteArray GstUriFromUrl(const QUrl& url);
void TagMessageReceived(GstMessage*);
void ErrorMessageReceived(GstMessage*);
void ElementMessageReceived(GstMessage*);
void StateChangedMessageReceived(GstMessage*);
void BufferingMessageReceived(GstMessage*);
void StreamStatusMessageReceived(GstMessage*);
QString ParseTag(GstTagList* list, const char* tag) const;
bool InitAudioBin();
GstElement* CreateDecodeBinFromString(const char* pipeline);
GstElement* CreateDecodeBinFromUrl(const QUrl& url);
void UpdateVolume();
void UpdateEqualizer();
void UpdateStereoBalance();
void SetOutputFormat(const QString& format);
bool ReplaceDecodeBin(GstElement* new_bin);
bool ReplaceDecodeBin(const QUrl& url);
void TransitionToNext();
// If the decodebin is special (ie. not really a uridecodebin) then it'll have
// a src pad immediately and we can link it after everything's created.
void MaybeLinkDecodeToAudio();
// Helper method to retrieve the audio format from a GstCaps object.
static QString GetAudioFormat(GstCaps* caps);
private slots:
void FaderTimelineFinished();
private:
static const int kGstStateTimeoutNanosecs;
static const int kFaderFudgeMsec;
static const int kEqBandCount;
static const int kEqBandFrequencies[];
static GstElementDeleter* sElementDeleter;
GstEngine* engine_;
// General settings for the pipeline
bool valid_;
QString sink_;
QVariant device_;
// These get called when there is a new audio buffer available
QList<BufferConsumer*> buffer_consumers_;
QMutex buffer_consumers_mutex_;
qint64 segment_start_;
bool segment_start_received_;
bool emit_track_ended_on_stream_start_;
bool emit_track_ended_on_time_discontinuity_;
qint64 last_buffer_offset_;
// Equalizer
bool eq_enabled_;
int eq_preamp_;
QList<int> eq_band_gains_;
// Stereo balance.
// From -1.0 - 1.0
// -1.0 is left, 1.0 is right.
float stereo_balance_;
// ReplayGain
bool rg_enabled_;
int rg_mode_;
float rg_preamp_;
bool rg_compression_;
// Buffering
quint64 buffer_duration_nanosec_;
int buffer_min_fill_;
bool buffering_;
bool mono_playback_;
int sample_rate_;
QString format_;
// The URL that is currently playing, and the URL that is to be preloaded
// when the current track is close to finishing.
MediaPlaybackRequest current_;
MediaPlaybackRequest next_;
// 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 multi-part
// 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.
QUrl 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 decodebin to be connected.
bool pipeline_is_initialised_;
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_;
int volume_percent_;
qreal volume_modifier_;
std::unique_ptr<QTimeLine> fader_;
QBasicTimer fader_fudge_timer_;
bool use_fudge_timer_;
// Bins
// uridecodebin ! audiobin
GstElement* uridecodebin_;
GstElement* audiobin_;
// Elements in the audiobin. See comments in Init()'s definition.
GstElement* queue_;
GstElement* audioconvert_;
GstElement* rgvolume_;
GstElement* rglimiter_;
GstElement* audioconvert2_;
GstElement* equalizer_preamp_;
GstElement* equalizer_;
GstElement* stereo_panorama_;
GstElement* volume_;
GstElement* audioscale_;
GstElement* audiosink_;
GstElement* capsfilter_;
// tee and request pads.
GstElement* tee_;
GstPad* tee_probe_pad_;
GstPad* tee_audio_pad_;
QThreadPool set_state_threadpool_;
GstSegment last_decodebin_segment_;
};
#endif // GSTENGINEPIPELINE_H