Clementine-audio-player-Mac.../src/engines/gstengine.cpp

772 lines
24 KiB
C++

/***************************************************************************
* Copyright (C) 2003-2005 by Mark Kretschmann <markey@web.de> *
* Copyright (C) 2005 by Jakub Stachowski <qbast@go2.pl> *
* Copyright (C) 2006 Paul Cifarelli <paul@cifarelli.net> *
* *
* This program 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 2 of the License, or *
* (at your option) any later version. *
* *
* This program 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 this program; if not, write to the *
* Free Software Foundation, Inc., *
* 51 Franklin Steet, Fifth Floor, Boston, MA 02111-1307, USA. *
***************************************************************************/
#define DEBUG_PREFIX "Gst-Engine"
#include "gstengine.h"
#include "gstequalizer.h"
#include "gstenginepipeline.h"
#include <math.h>
#include <unistd.h>
#include <vector>
#include <iostream>
#include <boost/bind.hpp>
#include <QTimer>
#include <QRegExp>
#include <QFile>
#include <QSettings>
#include <QtDebug>
#include <QCoreApplication>
#include <QTimeLine>
#include <QDir>
#include <gst/gst.h>
using std::vector;
using boost::shared_ptr;
const char* GstEngine::kSettingsGroup = "GstEngine";
const char* GstEngine::kAutoSink = "autoaudiosink";
GstEngine::GstEngine()
: Engine::Base(),
delayq_(g_queue_new()),
current_sample_(0),
equalizer_enabled_(false),
rg_enabled_(false),
rg_mode_(0),
rg_preamp_(0.0),
rg_compression_(true),
seek_timer_(new QTimer(this)),
timer_id_(-1)
{
seek_timer_->setSingleShot(true);
seek_timer_->setInterval(kSeekDelay);
connect(seek_timer_, SIGNAL(timeout()), SLOT(SeekNow()));
ReloadSettings();
}
GstEngine::~GstEngine() {
current_pipeline_.reset();
// Destroy scope delay queue
ClearScopeBuffers();
g_queue_free(delayq_);
// Save configuration
gst_deinit();
}
void GstEngine::SetEnv(const char *key, const QString &value) {
#ifdef Q_OS_WIN32
putenv(QString("%1=%2").arg(key, value).toLocal8Bit().constData());
#else
setenv(key, value.toLocal8Bit().constData(), 1);
#endif
}
bool GstEngine::Init() {
QString scanner_path;
QString plugin_path;
QString registry_filename;
// On windows and mac we bundle the gstreamer plugins with clementine
#if defined(Q_OS_DARWIN)
scanner_path = QCoreApplication::applicationDirPath() + "/../PlugIns/gst-plugin-scanner";
plugin_path = QCoreApplication::applicationDirPath() + "/../PlugIns/gstreamer";
#elif defined(Q_OS_WIN32)
plugin_path = QCoreApplication::applicationDirPath() + "/gstreamer-plugins";
#endif
#if defined(Q_OS_WIN32) || defined(Q_OS_DARWIN)
registry_filename = QString("%1/.config/%2/gst-registry-%3.bin").arg(
QDir::homePath(), QCoreApplication::organizationName(),
QCoreApplication::applicationVersion());
#endif
if (!scanner_path.isEmpty())
SetEnv("GST_PLUGIN_SCANNER", scanner_path);
if (!plugin_path.isEmpty()) {
SetEnv("GST_PLUGIN_PATH", plugin_path);
// Never load plugins from anywhere else.
SetEnv("GST_PLUGIN_SYSTEM_PATH", plugin_path);
}
if (!registry_filename.isEmpty()) {
SetEnv("GST_REGISTRY", registry_filename);
}
// GStreamer initialization
GError *err;
if ( !gst_init_check( NULL, NULL, &err ) ) {
qWarning("GStreamer could not be initialized");
return false;
}
return true;
}
void GstEngine::ReloadSettings() {
Engine::Base::ReloadSettings();
QSettings s;
s.beginGroup(kSettingsGroup);
sink_ = s.value("sink", kAutoSink).toString();
device_ = s.value("device").toString();
#ifdef Q_OS_WIN32
if (sink_ == kAutoSink) {
// HACK: Force the direct sound sink on Windows unless the user has
// explicitly chosen otherwise.
sink_ = "directsoundsink";
}
#endif
rg_enabled_ = s.value("rgenabled", false).toBool();
rg_mode_ = s.value("rgmode", 0).toInt();
rg_preamp_ = s.value("rgpreamp", 0.0).toDouble();
rg_compression_ = s.value("rgcompression", true).toBool();
}
bool GstEngine::CanDecode(const QUrl &url) {
// We had some bug reports claiming that video files cause crashes in canDecode(),
// so don't try to decode them
if ( url.path().toLower().endsWith( ".mov" ) ||
url.path().toLower().endsWith( ".avi" ) ||
url.path().toLower().endsWith( ".wmv" ) )
return false;
can_decode_success_ = false;
can_decode_last_ = false;
// Create the pipeline
shared_ptr<GstElement> pipeline(gst_pipeline_new("pipeline"),
boost::bind(gst_object_unref, _1));
if (!pipeline) return false;
GstElement* src = CreateElement("giosrc", pipeline.get()); if (!src) return false;
GstElement* bin = CreateElement("decodebin2", pipeline.get()); if (!bin) return false;
gst_element_link(src, bin);
g_signal_connect(G_OBJECT(bin), "new-decoded-pad", G_CALLBACK(CanDecodeNewPadCallback), this);
g_signal_connect(G_OBJECT(bin), "no-more-pads", G_CALLBACK(CanDecodeLastCallback), this);
// These handlers just print out errors to stderr
gst_bus_set_sync_handler(gst_pipeline_get_bus(GST_PIPELINE(pipeline.get())), CanDecodeBusCallbackSync, 0);
gst_bus_add_watch(gst_pipeline_get_bus(GST_PIPELINE(pipeline.get())), CanDecodeBusCallback, 0);
// Set the file we're testing
g_object_set(G_OBJECT(src), "location", url.toEncoded().constData(), NULL);
// Start the pipeline playing
gst_element_set_state(pipeline.get(), GST_STATE_PLAYING);
// Wait until found audio stream
int count = 0;
while (!can_decode_success_ && !can_decode_last_ && count < 100) {
count++;
usleep(1000);
}
// Stop playing
gst_element_set_state(pipeline.get(), GST_STATE_NULL);
return can_decode_success_;
}
void GstEngine::CanDecodeNewPadCallback(GstElement*, GstPad* pad, gboolean, gpointer self) {
GstEngine* instance = reinterpret_cast<GstEngine*>(self);
GstCaps* caps = gst_pad_get_caps(pad);
if (gst_caps_get_size(caps) > 0) {
GstStructure* str = gst_caps_get_structure(caps, 0);
if (g_strrstr(gst_structure_get_name( str ), "audio" ))
instance->can_decode_success_ = true;
}
gst_caps_unref(caps);
}
void GstEngine::CanDecodeLastCallback(GstElement*, gpointer self) {
GstEngine* instance = reinterpret_cast<GstEngine*>(self);
instance->can_decode_last_ = true;
}
void GstEngine::PrintGstError(GstMessage *msg) {
GError* error;
gchar* debugs;
gst_message_parse_error(msg, &error, &debugs);
qDebug() << error->message;
qDebug() << debugs;
g_error_free(error);
free(debugs);
}
GstBusSyncReply GstEngine::CanDecodeBusCallbackSync(GstBus*, GstMessage* msg, gpointer) {
if (GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ERROR)
PrintGstError(msg);
return GST_BUS_PASS;
}
gboolean GstEngine::CanDecodeBusCallback(GstBus*, GstMessage* msg, gpointer) {
if (GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ERROR)
PrintGstError(msg);
return GST_BUS_DROP;
}
uint GstEngine::position() const {
if (!current_pipeline_)
return 0;
return uint(current_pipeline_->position() / GST_MSECOND);
}
uint GstEngine::length() const {
if (!current_pipeline_)
return 0;
return uint(current_pipeline_->length() / GST_MSECOND);
}
Engine::State GstEngine::state() const {
if (!current_pipeline_)
return url_.isEmpty() ? Engine::Empty : Engine::Idle;
switch (current_pipeline_->state()) {
case GST_STATE_NULL: return Engine::Empty;
case GST_STATE_READY: return Engine::Idle;
case GST_STATE_PLAYING: return Engine::Playing;
case GST_STATE_PAUSED: return Engine::Paused;
default: return Engine::Empty;
}
}
void GstEngine::ConsumeBuffer(GstBuffer *buffer, GstEnginePipeline* pipeline) {
// Schedule this to run in the GUI thread. The buffer gets added to the
// queue and unreffed by UpdateScope.
if (!QMetaObject::invokeMethod(this, "AddBufferToScope",
Q_ARG(GstBuffer*, buffer),
Q_ARG(GstEnginePipeline*, pipeline))) {
qWarning() << "Failed to invoke AddBufferToScope on GstEngine";
}
}
void GstEngine::AddBufferToScope(GstBuffer* buf, GstEnginePipeline* pipeline) {
if (current_pipeline_.get() != pipeline) {
gst_buffer_unref(buf);
return;
}
g_queue_push_tail(delayq_, buf);
}
const Engine::Scope& GstEngine::scope() {
UpdateScope();
if (current_sample_ >= kScopeSize) {
// ok, we have a full buffer now, so give it to the scope
for (int i=0; i< kScopeSize; i++)
scope_[i] = current_scope_[i];
current_sample_ = 0;
}
return scope_;
}
void GstEngine::UpdateScope() {
typedef int16_t sampletype;
// prune the scope and get the current pos of the audio device
quint64 pos = PruneScope();
// head of the delay queue is the most delayed, so we work with that one
GstBuffer *buf = reinterpret_cast<GstBuffer *>( g_queue_peek_head(delayq_) );
if (!buf)
return;
// start time for this buffer
quint64 stime = GST_BUFFER_TIMESTAMP(buf);
// duration of the buffer...
quint64 dur = GST_BUFFER_DURATION(buf);
// therefore we can calculate the end time for the buffer
quint64 etime = stime + dur;
// determine the number of channels
GstStructure* structure = gst_caps_get_structure ( GST_BUFFER_CAPS( buf ), 0);
int channels = 2;
gst_structure_get_int (structure, "channels", &channels);
// scope does not support >2 channels
if (channels > 2)
return;
// if the audio device is playing this buffer now
if (pos <= stime || pos >= etime)
return;
// calculate the number of samples in the buffer
int sz = GST_BUFFER_SIZE(buf) / sizeof(sampletype);
// number of frames is the number of samples in each channel (frames like in the alsa sense)
int frames = sz / channels;
// find the offset into the buffer to the sample closest to where the audio device is playing
// it is the (time into the buffer cooresponding to the audio device pos) / (the sample rate)
// sample rate = duration of the buffer / number of frames in the buffer
// then we multiply by the number of channels to find the offset of the left channel sample
// of the frame in the buffer
int off = channels * (pos - stime) / (dur / frames);
// note that we are assuming 32 bit samples, but this should probably be generalized...
sampletype* data = reinterpret_cast<sampletype *>(GST_BUFFER_DATA(buf));
if (off >= sz) // better be...
return;
int i = off; // starting at offset
// loop while we fill the current buffer. If we need another buffer and one is available,
// get it and keep filling. If there are no more buffers available (not too likely)
// then leave everything in this state and wait until the next time the scope updates
while (buf && current_sample_ < kScopeSize && i < sz) {
for (int j = 0; j < channels && current_sample_ < kScopeSize; j++) {
current_scope_[current_sample_ ++] = data[i + j];
}
i+=channels; // advance to the next frame
if (i >= sz - 1) {
// here we are out of samples in the current buffer, so we get another one
buf = reinterpret_cast<GstBuffer *>( g_queue_pop_head(delayq_) );
gst_buffer_unref(buf);
buf = reinterpret_cast<GstBuffer *>( g_queue_peek_head(delayq_) );
if (buf) {
stime = GST_BUFFER_TIMESTAMP(buf);
dur = GST_BUFFER_DURATION(buf);
etime = stime + dur;
i = 0;
sz = GST_BUFFER_SIZE(buf) / sizeof(sampletype);
data = reinterpret_cast<sampletype *>(GST_BUFFER_DATA(buf));
}
}
}
}
void GstEngine::StartPreloading(const QUrl& url) {
if (autocrossfade_enabled_) {
// Have to create a new pipeline so we can crossfade between the two
preload_pipeline_ = CreatePipeline(url);
if (!preload_pipeline_)
return;
// We don't want to get metadata messages before the track starts playing -
// we reconnect this in GstEngine::Load
disconnect(preload_pipeline_.get(), SIGNAL(MetadataFound(Engine::SimpleMetaBundle)), this, 0);
preloaded_url_ = url;
preload_pipeline_->SetState(GST_STATE_PAUSED);
} else {
// No crossfading, so we can just queue the new URL in the existing
// pipeline and get gapless playback (hopefully)
if (current_pipeline_)
current_pipeline_->SetNextUrl(url);
}
}
bool GstEngine::Load(const QUrl& url, Engine::TrackChangeType change) {
Engine::Base::Load(url, change);
// Clementine just crashes when asked to load a file that doesn't exist on
// Windows, so check for that here. This is definitely the wrong place for
// this "fix"...
if (url.scheme() == "file" && !QFile::exists(url.toLocalFile()))
return false;
QUrl gst_url = url;
// It's a file:// url with a hostname set. QUrl::fromLocalFile does this
// when given a \\host\share\file path on Windows. Munge it back into a
// path that gstreamer will recognise.
if (url.scheme() == "file" && !url.host().isEmpty()) {
gst_url.setPath("//" + gst_url.host() + gst_url.path());
gst_url.setHost(QString());
}
const bool crossfade = current_pipeline_ &&
((crossfade_enabled_ && change == Engine::Manual) ||
(autocrossfade_enabled_ && change == Engine::Auto));
if (!crossfade && current_pipeline_ && current_pipeline_->url() == gst_url &&
change == Engine::Auto) {
// We're not crossfading, and the pipeline is already playing the URI we
// want, so just do nothing.
return true;
}
shared_ptr<GstEnginePipeline> pipeline;
if (preload_pipeline_ && preloaded_url_ == gst_url) {
pipeline = preload_pipeline_;
connect(preload_pipeline_.get(),
SIGNAL(MetadataFound(Engine::SimpleMetaBundle)),
SLOT(NewMetaData(Engine::SimpleMetaBundle)));
} else {
pipeline = CreatePipeline(gst_url);
if (!pipeline)
return false;
}
if (crossfade)
StartFadeout();
current_pipeline_ = pipeline;
preload_pipeline_.reset();
SetVolume(volume_);
SetEqualizerEnabled(equalizer_enabled_);
SetEqualizerParameters(equalizer_preamp_, equalizer_gains_);
// Maybe fade in this track
if (crossfade)
current_pipeline_->StartFader(fadeout_duration_, QTimeLine::Forward);
return true;
}
void GstEngine::StartFadeout() {
fadeout_pipeline_ = current_pipeline_;
disconnect(fadeout_pipeline_.get(), 0, 0, 0);
fadeout_pipeline_->RemoveAllBufferConsumers();
ClearScopeBuffers();
fadeout_pipeline_->StartFader(fadeout_duration_, QTimeLine::Backward);
connect(fadeout_pipeline_.get(), SIGNAL(FaderFinished()), SLOT(FadeoutFinished()));
}
bool GstEngine::Play( uint offset ) {
// Try to play input pipeline; if fails, destroy input bin
if (!current_pipeline_->SetState(GST_STATE_PLAYING)) {
qWarning() << "Could not set thread to PLAYING.";
current_pipeline_.reset();
return false;
}
// If "Resume playback on start" is enabled, we must seek to the last position
if (offset) Seek(offset);
if (timer_id_ != -1)
killTimer(timer_id_);
timer_id_ = startTimer(kTimerInterval);
current_sample_ = 0;
emit StateChanged(Engine::Playing);
return true;
}
void GstEngine::Stop() {
killTimer(timer_id_);
timer_id_ = -1;
url_ = QUrl(); // To ensure we return Empty from state()
if (fadeout_enabled_ && current_pipeline_)
StartFadeout();
current_pipeline_.reset();
emit StateChanged(Engine::Empty);
}
void GstEngine::FadeoutFinished() {
fadeout_pipeline_.reset();
}
void GstEngine::Pause() {
if (!current_pipeline_)
return;
if ( current_pipeline_->state() == GST_STATE_PLAYING ) {
current_pipeline_->SetState(GST_STATE_PAUSED);
emit StateChanged(Engine::Paused);
}
}
void GstEngine::Unpause() {
if (!current_pipeline_)
return;
if ( current_pipeline_->state() == GST_STATE_PAUSED ) {
current_pipeline_->SetState(GST_STATE_PLAYING);
emit StateChanged(Engine::Playing);
}
}
void GstEngine::Seek(uint ms) {
if (!current_pipeline_)
return;
seek_pos_ = ms;
waiting_to_seek_ = true;
if (!seek_timer_->isActive()) {
SeekNow();
seek_timer_->start(); // Stop us from seeking again for a little while
}
}
void GstEngine::SeekNow() {
if (!waiting_to_seek_) return;
waiting_to_seek_ = false;
if (!current_pipeline_)
return;
if (current_pipeline_->Seek(seek_pos_ * GST_MSECOND))
ClearScopeBuffers();
else
qDebug() << "Seek failed";
}
void GstEngine::SetEqualizerEnabled(bool enabled) {
equalizer_enabled_= enabled;
if (current_pipeline_)
current_pipeline_->SetEqualizerEnabled(enabled);
}
void GstEngine::SetEqualizerParameters(int preamp, const QList<int>& band_gains) {
equalizer_preamp_ = preamp;
equalizer_gains_ = band_gains;
if (current_pipeline_)
current_pipeline_->SetEqualizerParams(preamp, band_gains);
}
void GstEngine::SetVolumeSW( uint percent ) {
if (current_pipeline_)
current_pipeline_->SetVolume(percent);
}
void GstEngine::timerEvent( QTimerEvent* ) {
// keep the scope from building while we are not visible
// this is why the timer must run as long as we are playing, and not just when
// we are fading
PruneScope();
// Emit TrackAboutToEnd when we're a few seconds away from finishing
if (current_pipeline_) {
const qint64 nanosec_position = current_pipeline_->position();
const qint64 nanosec_length = current_pipeline_->length();
const qint64 remaining = (nanosec_length - nanosec_position) / 1000000;
const qint64 fudge = 100; // Mmm fudge
const qint64 gap = autocrossfade_enabled_ ? fadeout_duration_ : kPreloadGap;
if (nanosec_length > 0 && remaining < gap + fudge)
EmitAboutToEnd();
}
}
void GstEngine::HandlePipelineError(const QString& message) {
qWarning() << "Gstreamer error:" << message;
current_pipeline_.reset();
emit Error(message);
emit StateChanged(Engine::Empty);
}
void GstEngine::EndOfStreamReached(bool has_next_track) {
if (!has_next_track)
current_pipeline_.reset();
ClearScopeBuffers();
emit TrackEnded();
}
void GstEngine::NewMetaData(const Engine::SimpleMetaBundle& bundle) {
emit MetaData(bundle);
}
GstElement* GstEngine::CreateElement(
const QString& factoryName, GstElement* bin, const QString& name ) {
GstElement* element =
gst_element_factory_make(
factoryName.toAscii().constData(),
name.isNull() ? factoryName.toAscii().constData() : name.toAscii().constData() );
if ( element ) {
if ( bin ) gst_bin_add( GST_BIN( bin ), element );
} else {
emit Error(QString("GStreamer could not create the element: %1. "
"Please make sure that you have installed all necessary GStreamer plugins (e.g. OGG and MP3)").arg( factoryName ) );
gst_object_unref( GST_OBJECT( bin ) );
}
return element;
}
GstEngine::PluginDetailsList
GstEngine::GetPluginList(const QString& classname) const {
PluginDetailsList ret;
GstRegistry* registry = gst_registry_get_default();
GList* const features =
gst_registry_get_feature_list(registry, GST_TYPE_ELEMENT_FACTORY);
GList* p = features;
while (p) {
GstElementFactory* factory = GST_ELEMENT_FACTORY(p->data);
if (QString(factory->details.klass).contains(classname)) {
PluginDetails details;
details.name = QString::fromUtf8(GST_PLUGIN_FEATURE_NAME(p->data));
details.long_name = QString::fromUtf8(factory->details.longname);
details.description = QString::fromUtf8(factory->details.description);
details.author = QString::fromUtf8(factory->details.author);
ret << details;
}
p = g_list_next(p);
}
gst_plugin_feature_list_free(features);
return ret;
}
shared_ptr<GstEnginePipeline> GstEngine::CreatePipeline(const QUrl& url) {
shared_ptr<GstEnginePipeline> ret(new GstEnginePipeline(this));
ret->set_output_device(sink_, device_);
ret->set_replaygain(rg_enabled_, rg_mode_, rg_preamp_, rg_compression_);
ret->AddBufferConsumer(this);
foreach (BufferConsumer* consumer, buffer_consumers_)
ret->AddBufferConsumer(consumer);
connect(ret.get(), SIGNAL(EndOfStreamReached(bool)), SLOT(EndOfStreamReached(bool)));
connect(ret.get(), SIGNAL(Error(QString)), SLOT(HandlePipelineError(QString)));
connect(ret.get(), SIGNAL(MetadataFound(Engine::SimpleMetaBundle)),
SLOT(NewMetaData(Engine::SimpleMetaBundle)));
connect(ret.get(), SIGNAL(destroyed()), SLOT(ClearScopeBuffers()));
if (!ret->Init(url))
ret.reset();
return ret;
}
qint64 GstEngine::PruneScope() {
if (!current_pipeline_)
return 0;
// get the position playing in the audio device
qint64 pos = current_pipeline_->position();
GstBuffer *buf = 0;
quint64 etime = 0;
// free up the buffers that the audio device has advanced past already
do {
// most delayed buffers are at the head of the queue
buf = reinterpret_cast<GstBuffer *>( g_queue_peek_head(delayq_) );
if (buf) {
// the start time of the buffer
quint64 stime = GST_BUFFER_TIMESTAMP(buf);
// the duration of the buffer
quint64 dur = GST_BUFFER_DURATION(buf);
// therefore we can calculate the end time of the buffer
etime = stime + dur;
// purge this buffer if the pos is past the end time of the buffer
if (pos > qint64(etime)) {
g_queue_pop_head(delayq_);
gst_buffer_unref(buf);
}
}
} while (buf && pos > qint64(etime));
return pos;
}
void GstEngine::ClearScopeBuffers() {
// just free them all
while (g_queue_get_length(delayq_)) {
GstBuffer* buf = reinterpret_cast<GstBuffer *>( g_queue_pop_head(delayq_) );
gst_buffer_unref(buf);
}
}
bool GstEngine::DoesThisSinkSupportChangingTheOutputDeviceToAUserEditableString(const QString &name) {
return (name == "alsasink" || name == "osssink" || name == "pulsesink");
}
void GstEngine::AddBufferConsumer(BufferConsumer *consumer) {
buffer_consumers_ << consumer;
if (current_pipeline_)
current_pipeline_->AddBufferConsumer(consumer);
}
void GstEngine::RemoveBufferConsumer(BufferConsumer *consumer) {
buffer_consumers_.removeAll(consumer);
if (current_pipeline_)
current_pipeline_->RemoveBufferConsumer(consumer);
}
int GstEngine::AddBackgroundStream(const QUrl& url) {
shared_ptr<GstEnginePipeline> pipeline = CreatePipeline(url);
if (!pipeline) {
return -1;
}
pipeline->SetVolume(30);
// We don't want to get metadata messages or end notifications.
disconnect(pipeline.get(), SIGNAL(MetadataFound(Engine::SimpleMetaBundle)), this, 0);
disconnect(pipeline.get(), SIGNAL(EndOfStreamReached(bool)), this, 0);
connect(pipeline.get(), SIGNAL(EndOfStreamReached(bool)), SLOT(BackgroundStreamFinished()));
if (!pipeline->SetState(GST_STATE_PLAYING)) {
qWarning() << "Could not set thread to PLAYING.";
pipeline.reset();
return -1;
}
pipeline->SetNextUrl(url);
int stream_id = next_background_stream_id_++;
background_streams_[stream_id] = pipeline;
return stream_id;
}
void GstEngine::StopBackgroundStream(int id) {
background_streams_.remove(id); // Removes last shared_ptr reference.
}
void GstEngine::BackgroundStreamFinished() {
GstEnginePipeline* pipeline = qobject_cast<GstEnginePipeline*>(sender());
pipeline->SetNextUrl(pipeline->url());
}