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

/***************************************************************************
 *   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 <QTimer>
#include <QRegExp>
#include <QFile>
#include <QMessageBox>
#include <QSettings>
#include <QtDebug>
#include <QCoreApplication>
#include <QTimeLine>

#include <gst/gst.h>
#include <iostream>


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),
    shutdown_(false),
    can_decode_pipeline_(NULL),
    can_decode_src_(NULL),
    can_decode_bin_(NULL)
{
  ReloadSettings();
}

GstEngine::~GstEngine() {
  current_pipeline_.reset();

  if (can_decode_pipeline_)
    gst_object_unref(GST_OBJECT(can_decode_pipeline_));

  // Destroy scope delay queue
  ClearScopeQ();
  g_queue_free(delayq_);

  // Save configuration
  gst_deinit();
}


bool GstEngine::init() {
  // GStreamer initialization
  GError *err;
  if ( !gst_init_check( NULL, NULL, &err ) ) {
    qWarning("GStreamer could not be initialized");
    return false;
  }

#ifdef Q_OS_WIN32
  // Set the plugin path on windows
  GstRegistry* registry = gst_registry_get_default();
  gst_registry_add_path(registry, QString(
      QCoreApplication::applicationDirPath() + "/gstreamer-plugins").toLocal8Bit().constData());
#endif

  return true;
}

void GstEngine::ReloadSettings() {
  QSettings s;
  s.beginGroup(kSettingsGroup);

  sink_ = s.value("sink", kAutoSink).toString();
  device_ = s.value("device").toString();
  fadeout_enabled_ = s.value("FadeoutEnabled", true).toBool();
  fadeout_duration_ = s.value("FadeoutDuration", 2000).toInt();
}


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
  if (!can_decode_pipeline_) {
    can_decode_pipeline_ = CreateElement("pipeline");
    can_decode_src_ = CreateElement("giosrc", can_decode_pipeline_);
    can_decode_bin_ = CreateElement("decodebin", can_decode_pipeline_);

    gst_element_link(can_decode_src_, can_decode_bin_);
    g_signal_connect(G_OBJECT(can_decode_bin_), "new-decoded-pad", G_CALLBACK(CanDecodeNewPadCallback), this);
    g_signal_connect(G_OBJECT(can_decode_bin_), "no-more-pads", G_CALLBACK(CanDecodeLastCallback), this);
  }

  // Set the file we're testing
  g_object_set(G_OBJECT(can_decode_src_), "location", url.toEncoded().constData(), NULL);

  // Start the pipeline playing
  gst_element_set_state(can_decode_pipeline_, 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(can_decode_pipeline_, GST_STATE_NULL);

  return can_decode_success_;
}

void GstEngine::CanDecodeNewPadCallback(GstElement*, GstPad* pad, gboolean, gpointer self) {
  GstEngine* instance = static_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 = static_cast<GstEngine*>(self);
  instance->can_decode_last_ = true;
}


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 m_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::NewBuffer(GstBuffer* buf) {
  g_queue_push_tail(delayq_, buf);
}

const Engine::Scope& GstEngine::scope() {
  UpdateScope();

  if (current_sample_ >= SCOPESIZE) {
    // ok, we have a full buffer now, so give it to the scope
    for (int i=0; i< SCOPESIZE; i++)
      m_scope[i] = current_scope_[i];
    current_sample_ = 0;
  }

  return m_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_ < SCOPESIZE && i < sz) {
    for (int j = 0; j < channels && current_sample_ < SCOPESIZE; 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));
      }
    }
  }
}

bool GstEngine::load(const QUrl& url, bool stream) {
  Engine::Base::load( url, stream );

  shared_ptr<GstEnginePipeline> pipeline(CreatePipeline(url));
  if (!pipeline)
    return false;

  current_pipeline_ = pipeline;

  setVolume(m_volume);
  setEqualizerEnabled(equalizer_enabled_);
  setEqualizerParameters(equalizer_preamp_, equalizer_gains_);
  return true;
}


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;
  }

  // Stop any active fadeout
  fadeout_pipeline_.reset();

  // If "Resume playback on start" is enabled, we must seek to the last position
  if (offset) seek(offset);

  current_sample_ = 0;
  startTimer(kTimerInterval);
  emit stateChanged(Engine::Playing);
  return true;
}


void GstEngine::stop() {
  m_url = QUrl(); // To ensure we return Empty from state()

  if (fadeout_enabled_) {
    fadeout_pipeline_ = current_pipeline_;
    disconnect(fadeout_pipeline_.get(), 0, 0, 0);
    ClearScopeQ();

    QTimeLine* fadeout = new QTimeLine(fadeout_duration_, this);
    connect(fadeout, SIGNAL(valueChanged(qreal)), fadeout_pipeline_.get(), SLOT(SetVolumeModifier(qreal)));
    connect(fadeout, SIGNAL(finished()), SLOT(FadeoutFinished()));
    connect(fadeout_pipeline_.get(), SIGNAL(destroyed()), fadeout, SLOT(deleteLater()));
    fadeout->setDirection(QTimeLine::Backward);
    fadeout->start();
  }

  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;

  if (current_pipeline_->Seek(ms * GST_MSECOND))
    ClearScopeQ();
  else
    qDebug() << "Seek failed";

  // ??
  //gst_element_get_state(gst_pipeline_, NULL, NULL, 100*GST_MSECOND);
}

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();

  // *** Volume fading ***

  // Are we currently fading?
  /*if ( fade_value_ > 0.0 ) {
    // TODO
    //m_fadeValue -= ( AmarokConfig::fadeoutLength() ) ?  1.0 / AmarokConfig::fadeoutLength() * TIMER_INTERVAL : 1.0;
    fade_value_ -= 1.0;

    // Fade finished?
    if ( fade_value_ <= 0.0 ) {
      // Fade transition has finished, stop playback
      qDebug() << "[Gst-Engine] Fade-out finished.";
      DestroyPipeline();
      //killTimers();
    }
    setVolume( volume() );
  }*/
}


/////////////////////////////////////////////////////////////////////////////////////
// PRIVATE SLOTS
/////////////////////////////////////////////////////////////////////////////////////

void GstEngine::HandlePipelineError(const QString& message) {
  qDebug() << "Gstreamer error:" << message;

  current_pipeline_.reset();
}


void GstEngine::EndOfStreamReached() {
  current_pipeline_.reset();
  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 {
    QMessageBox::critical( 0, "Error",
                           QString("<h3>GStreamer could not create the element: <i>%1</i></h3> "
                                   "<p>Please make sure that you have installed all necessary GStreamer plugins (e.g. OGG and MP3), and run <i>'gst-register'</i> afterwards.</p>"
                                   "<p>For further assistance consult the GStreamer manual, and join #gstreamer on irc.freenode.net.</p>" ).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* features =
      gst_registry_get_feature_list(registry, GST_TYPE_ELEMENT_FACTORY);

  while (features) {
    GstElementFactory* factory = GST_ELEMENT_FACTORY(features->data);
    if (QString(factory->details.klass).contains(classname)) {
      PluginDetails details;
      details.name = QString::fromUtf8(GST_PLUGIN_FEATURE_NAME(features->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;
    }
    features = g_list_next ( features );
  }

  gst_plugin_feature_list_free(features);
  return ret;
}

shared_ptr<GstEnginePipeline> GstEngine::CreatePipeline(const QUrl& url) {
  shared_ptr<GstEnginePipeline> ret(new GstEnginePipeline);
  ret->set_forwards_buffers(true);
  ret->set_output_device(sink_, device_);

  connect(ret.get(), SIGNAL(EndOfStreamReached()), SLOT(EndOfStreamReached()));
  connect(ret.get(), SIGNAL(BufferFound(GstBuffer*)), SLOT(NewBuffer(GstBuffer*)));
  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(ClearScopeQ()));

  if (!ret->Init(url))
    ret.reset();

  return ret;
}

qint64 GstEngine::PruneScope() {
  if (!current_pipeline_)
    return 0;

  // get the position playing in the audio device
  gint64 pos = current_pipeline_->position();

  GstBuffer *buf = 0;
  quint64 etime;

  // 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::ClearScopeQ() {
  // 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");
}