void GstEnginePipeline::TransitionToNext() {
GstElement* old_decode_bin = uridecodebin_;
ignore_tags_ = true;
ReplaceDecodeBin(next_url_);
gst_element_set_state(uridecodebin_, GST_STATE_PLAYING);
MaybeLinkDecodeToAudio();
url_ = next_url_;
end_offset_nanosec_ = next_end_offset_nanosec_;
next_url_ = QUrl();
next_beginning_offset_nanosec_ = 0;
next_end_offset_nanosec_ = 0;
// This function gets called when the source has been drained, even if the
// song hasn't finished playing yet. We'll get a new segment when it really
// does finish, so emit TrackEnded then.
emit_track_ended_on_segment_start_ = true;
// This has to happen *after* the gst_element_set_state on the new bin to
// fix an occasional race condition deadlock.
sElementDeleter->DeleteElementLater(old_decode_bin);
ignore_tags_ = false;
}
bool GstEnginePipeline::Init() {
// Here we create all the parts of the gstreamer pipeline - from the source
// to the sink. The parts of the pipeline are split up into bins:
// uri decode bin -> audio bin
// The uri decode bin is a gstreamer builtin that automatically picks the
// right type of source and decoder for the URI.
// The audio bin gets created here and contains:
// queue ! audioconvert ! <caps32>
// ! ( rgvolume ! rglimiter ! audioconvert2 ) ! tee
// rgvolume and rglimiter are only created when replaygain is enabled.
// After the tee the pipeline splits. One split is converted to 16-bit int
// samples for the scope, the other is kept as float32 and sent to the
// speaker.
// tee1 ! probe_queue ! probe_converter ! <caps16> ! probe_sink
// tee2 ! audio_queue ! equalizer_preamp ! equalizer ! volume ! audioscale
// ! convert ! audiosink
// Audio bin
audiobin_ = gst_bin_new("audiobin");
gst_bin_add(GST_BIN(pipeline_), audiobin_);
// Create the sink
if (!(audiosink_ = engine_->CreateElement(sink_, audiobin_))) return false;
if (g_object_class_find_property(G_OBJECT_GET_CLASS(audiosink_), "device") &&
!device_.toString().isEmpty()) {
switch (device_.type()) {
case QVariant::Int:
g_object_set(G_OBJECT(audiosink_),
"device", device_.toInt(),
nullptr);
break;
case QVariant::String:
g_object_set(G_OBJECT(audiosink_),
"device", device_.toString().toUtf8().constData(),
nullptr);
break;
#ifdef Q_OS_WIN32
case QVariant::ByteArray: {
GUID guid = QUuid(device_.toByteArray());
g_object_set(G_OBJECT(audiosink_),
"device", &guid,
nullptr);
break;
}
#endif // Q_OS_WIN32
default:
qLog(Warning) << "Unknown device type" << device_;
break;
}
}
// Create all the other elements
GstElement* tee, *probe_queue, *probe_converter, *probe_sink, *audio_queue,
*convert;
queue_ = engine_->CreateElement("queue2", audiobin_);
audioconvert_ = engine_->CreateElement("audioconvert", audiobin_);
tee = engine_->CreateElement("tee", audiobin_);
probe_queue = engine_->CreateElement("queue", audiobin_);
probe_converter = engine_->CreateElement("audioconvert", audiobin_);
probe_sink = engine_->CreateElement("fakesink", audiobin_);
audio_queue = engine_->CreateElement("queue", audiobin_);
equalizer_preamp_ = engine_->CreateElement("volume", audiobin_);
equalizer_ = engine_->CreateElement("equalizer-nbands", audiobin_);
stereo_panorama_ = engine_->CreateElement("audiopanorama", audiobin_);
volume_ = engine_->CreateElement("volume", audiobin_);
audioscale_ = engine_->CreateElement("audioresample", audiobin_);
convert = engine_->CreateElement("audioconvert", audiobin_);
if (!queue_ || !audioconvert_ || !tee || !probe_queue || !probe_converter ||
!probe_sink || !audio_queue || !equalizer_preamp_ || !equalizer_ ||
!stereo_panorama_ || !volume_ || !audioscale_ || !convert) {
return false;
}
// Create the replaygain elements if it's enabled. event_probe is the
// audioconvert element we attach the probe to, which will change depending
// on whether replaygain is enabled. convert_sink is the element after the
// first audioconvert, which again will change.
GstElement* event_probe = audioconvert_;
GstElement* convert_sink = tee;
if (rg_enabled_) {
rgvolume_ = engine_->CreateElement("rgvolume", audiobin_);
rglimiter_ = engine_->CreateElement("rglimiter", audiobin_);
audioconvert2_ = engine_->CreateElement("audioconvert", audiobin_);
event_probe = audioconvert2_;
convert_sink = rgvolume_;
if (!rgvolume_ || !rglimiter_ || !audioconvert2_) {
return false;
}
// Set replaygain settings
g_object_set(G_OBJECT(rgvolume_), "album-mode", rg_mode_, nullptr);
g_object_set(G_OBJECT(rgvolume_), "pre-amp", double(rg_preamp_), nullptr);
g_object_set(G_OBJECT(rglimiter_), "enabled", int(rg_compression_),
nullptr);
}
// Create a pad on the outside of the audiobin and connect it to the pad of
// the first element.
GstPad* pad = gst_element_get_static_pad(queue_, "sink");
gst_element_add_pad(audiobin_, gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
// Add a data probe on the src pad of the audioconvert element for our scope.
// We do it here because we want pre-equalized and pre-volume samples
// so that our visualization are not be affected by them.
pad = gst_element_get_static_pad(event_probe, "src");
gst_pad_add_event_probe(pad, G_CALLBACK(EventHandoffCallback), this);
gst_object_unref(pad);
// Configure the fakesink properly
g_object_set(G_OBJECT(probe_sink), "sync", TRUE, nullptr);
// Set the equalizer bands
g_object_set(G_OBJECT(equalizer_), "num-bands", 10, nullptr);
int last_band_frequency = 0;
for (int i = 0; i < kEqBandCount; ++i) {
GstObject* band =
gst_child_proxy_get_child_by_index(GST_CHILD_PROXY(equalizer_), i);
const float frequency = kEqBandFrequencies[i];
const float bandwidth = frequency - last_band_frequency;
last_band_frequency = frequency;
g_object_set(G_OBJECT(band), "freq", frequency, "bandwidth", bandwidth,
"gain", 0.0f, nullptr);
g_object_unref(G_OBJECT(band));
}
// Set the stereo balance.
g_object_set(G_OBJECT(stereo_panorama_), "panorama", stereo_balance_,
nullptr);
// Set the buffer duration. We set this on this queue instead of the
// decode bin (in ReplaceDecodeBin()) because setting it on the decode bin
// only affects network sources.
// Disable the default buffer and byte limits, so we only buffer based on
// time.
g_object_set(G_OBJECT(queue_), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(queue_), "max-size-bytes", 0, nullptr);
g_object_set(G_OBJECT(queue_), "max-size-time", buffer_duration_nanosec_,
nullptr);
g_object_set(G_OBJECT(queue_), "low-percent", buffer_min_fill_, nullptr);
if (buffer_duration_nanosec_ > 0) {
g_object_set(G_OBJECT(queue_), "use-buffering", true, nullptr);
}
gst_element_link(queue_, audioconvert_);
// Create the caps to put in each path in the tee. The scope path gets 16-bit
// ints and the audiosink path gets float32.
GstCaps* caps16 =
gst_caps_new_simple("audio/x-raw-int", "width", G_TYPE_INT, 16, "signed",
G_TYPE_BOOLEAN, true, nullptr);
GstCaps* caps32 = gst_caps_new_simple("audio/x-raw-float", "width",
G_TYPE_INT, 32, nullptr);
if (mono_playback_) {
gst_caps_set_simple(caps32, "channels", G_TYPE_INT, 1, nullptr);
}
// Link the elements with special caps
gst_element_link_filtered(probe_converter, probe_sink, caps16);
gst_element_link_filtered(audioconvert_, convert_sink, caps32);
gst_caps_unref(caps16);
gst_caps_unref(caps32);
// Link the outputs of tee to the queues on each path.
gst_pad_link(gst_element_get_request_pad(tee, "src%d"),
gst_element_get_static_pad(probe_queue, "sink"));
gst_pad_link(gst_element_get_request_pad(tee, "src%d"),
gst_element_get_static_pad(audio_queue, "sink"));
// Link replaygain elements if enabled.
if (rg_enabled_) {
gst_element_link_many(rgvolume_, rglimiter_, audioconvert2_, tee, nullptr);
}
// Link everything else.
gst_element_link(probe_queue, probe_converter);
gst_element_link_many(audio_queue, equalizer_preamp_, equalizer_,
stereo_panorama_, volume_, audioscale_, convert,
audiosink_, nullptr);
// Add probes and handlers.
gst_pad_add_buffer_probe(gst_element_get_static_pad(probe_converter, "src"),
G_CALLBACK(HandoffCallback), this);
gst_bus_set_sync_handler(gst_pipeline_get_bus(GST_PIPELINE(pipeline_)),
BusCallbackSync, this);
bus_cb_id_ = gst_bus_add_watch(gst_pipeline_get_bus(GST_PIPELINE(pipeline_)),
BusCallback, this);
MaybeLinkDecodeToAudio();
return true;
}
bool GstEnginePipeline::Init() {
// Here we create all the parts of the gstreamer pipeline - from the source
// to the sink. The parts of the pipeline are split up into bins:
// uri decode bin -> audio bin
// The uri decode bin is a gstreamer builtin that automatically picks the
// right type of source and decoder for the URI.
// The audio bin gets created here and contains:
// queue ! audioconvert ! <caps32>
// ! ( rgvolume ! rglimiter ! audioconvert2 ) ! tee
// rgvolume and rglimiter are only created when replaygain is enabled.
// After the tee the pipeline splits. One split is converted to 16-bit int
// samples for the scope, the other is kept as float32 and sent to the
// speaker.
// tee1 ! probe_queue ! probe_converter ! <caps16> ! probe_sink
// tee2 ! audio_queue ! equalizer_preamp ! equalizer ! volume ! audioscale
// ! convert ! audiosink
gst_segment_init(&last_decodebin_segment_, GST_FORMAT_TIME);
// Audio bin
audiobin_ = gst_bin_new("audiobin");
gst_bin_add(GST_BIN(pipeline_), audiobin_);
// Create the sink
if (!(audiosink_ = engine_->CreateElement(sink_, audiobin_))) return false;
if (g_object_class_find_property(G_OBJECT_GET_CLASS(audiosink_), "device") &&
!device_.toString().isEmpty()) {
switch (device_.type()) {
case QVariant::Int:
g_object_set(G_OBJECT(audiosink_), "device", device_.toInt(), nullptr);
break;
case QVariant::LongLong:
g_object_set(G_OBJECT(audiosink_), "device", device_.toLongLong(),
nullptr);
break;
case QVariant::String:
g_object_set(G_OBJECT(audiosink_), "device",
device_.toString().toUtf8().constData(), nullptr);
break;
case QVariant::ByteArray: {
g_object_set(G_OBJECT(audiosink_), "device",
device_.toByteArray().constData(), nullptr);
break;
}
default:
qLog(Warning) << "Unknown device type" << device_;
break;
}
}
// Create all the other elements
GstElement* tee, *probe_queue, *probe_converter, *probe_sink, *audio_queue,
*convert;
queue_ = engine_->CreateElement("queue2", audiobin_);
audioconvert_ = engine_->CreateElement("audioconvert", audiobin_);
tee = engine_->CreateElement("tee", audiobin_);
probe_queue = engine_->CreateElement("queue2", audiobin_);
probe_converter = engine_->CreateElement("audioconvert", audiobin_);
probe_sink = engine_->CreateElement("fakesink", audiobin_);
audio_queue = engine_->CreateElement("queue", audiobin_);
equalizer_preamp_ = engine_->CreateElement("volume", audiobin_);
equalizer_ = engine_->CreateElement("equalizer-nbands", audiobin_);
stereo_panorama_ = engine_->CreateElement("audiopanorama", audiobin_);
volume_ = engine_->CreateElement("volume", audiobin_);
audioscale_ = engine_->CreateElement("audioresample", audiobin_);
convert = engine_->CreateElement("audioconvert", audiobin_);
if (!queue_ || !audioconvert_ || !tee || !probe_queue || !probe_converter ||
!probe_sink || !audio_queue || !equalizer_preamp_ || !equalizer_ ||
!stereo_panorama_ || !volume_ || !audioscale_ || !convert) {
return false;
}
// Create the replaygain elements if it's enabled. event_probe is the
// audioconvert element we attach the probe to, which will change depending
// on whether replaygain is enabled. convert_sink is the element after the
// first audioconvert, which again will change.
GstElement* event_probe = audioconvert_;
GstElement* convert_sink = tee;
if (rg_enabled_) {
rgvolume_ = engine_->CreateElement("rgvolume", audiobin_);
rglimiter_ = engine_->CreateElement("rglimiter", audiobin_);
audioconvert2_ = engine_->CreateElement("audioconvert", audiobin_);
event_probe = audioconvert2_;
convert_sink = rgvolume_;
if (!rgvolume_ || !rglimiter_ || !audioconvert2_) {
return false;
}
// Set replaygain settings
g_object_set(G_OBJECT(rgvolume_), "album-mode", rg_mode_, nullptr);
g_object_set(G_OBJECT(rgvolume_), "pre-amp", double(rg_preamp_), nullptr);
g_object_set(G_OBJECT(rglimiter_), "enabled", int(rg_compression_),
nullptr);
}
// Create a pad on the outside of the audiobin and connect it to the pad of
// the first element.
GstPad* pad = gst_element_get_static_pad(queue_, "sink");
gst_element_add_pad(audiobin_, gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
// Add a data probe on the src pad of the audioconvert element for our scope.
// We do it here because we want pre-equalized and pre-volume samples
// so that our visualization are not be affected by them.
pad = gst_element_get_static_pad(event_probe, "src");
gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_EVENT_UPSTREAM,
&EventHandoffCallback, this, NULL);
gst_object_unref(pad);
// Configure the fakesink properly
g_object_set(G_OBJECT(probe_sink), "sync", TRUE, nullptr);
// Setting the equalizer bands:
//
// GStreamer's GstIirEqualizerNBands sets up shelve filters for the first and
// last bands as corner cases. That was causing the "inverted slider" bug.
// As a workaround, we create two dummy bands at both ends of the spectrum.
// This causes the actual first and last adjustable bands to be
// implemented using band-pass filters.
g_object_set(G_OBJECT(equalizer_), "num-bands", 10 + 2, nullptr);
// Dummy first band (bandwidth 0, cutting below 20Hz):
GstObject* first_band = GST_OBJECT(
gst_child_proxy_get_child_by_index(GST_CHILD_PROXY(equalizer_), 0));
g_object_set(G_OBJECT(first_band), "freq", 20.0, "bandwidth", 0, "gain", 0.0f,
nullptr);
g_object_unref(G_OBJECT(first_band));
// Dummy last band (bandwidth 0, cutting over 20KHz):
GstObject* last_band = GST_OBJECT(gst_child_proxy_get_child_by_index(
GST_CHILD_PROXY(equalizer_), kEqBandCount + 1));
g_object_set(G_OBJECT(last_band), "freq", 20000.0, "bandwidth", 0, "gain",
0.0f, nullptr);
g_object_unref(G_OBJECT(last_band));
int last_band_frequency = 0;
for (int i = 0; i < kEqBandCount; ++i) {
const int index_in_eq = i + 1;
GstObject* band = GST_OBJECT(gst_child_proxy_get_child_by_index(
GST_CHILD_PROXY(equalizer_), index_in_eq));
const float frequency = kEqBandFrequencies[i];
const float bandwidth = frequency - last_band_frequency;
last_band_frequency = frequency;
g_object_set(G_OBJECT(band), "freq", frequency, "bandwidth", bandwidth,
"gain", 0.0f, nullptr);
g_object_unref(G_OBJECT(band));
}
// Set the stereo balance.
g_object_set(G_OBJECT(stereo_panorama_), "panorama", stereo_balance_,
nullptr);
// Set the buffer duration. We set this on this queue instead of the
// decode bin (in ReplaceDecodeBin()) because setting it on the decode bin
// only affects network sources.
// Disable the default buffer and byte limits, so we only buffer based on
// time.
g_object_set(G_OBJECT(queue_), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(queue_), "max-size-bytes", 0, nullptr);
g_object_set(G_OBJECT(queue_), "max-size-time", buffer_duration_nanosec_,
nullptr);
g_object_set(G_OBJECT(queue_), "low-percent", buffer_min_fill_, nullptr);
if (buffer_duration_nanosec_ > 0) {
g_object_set(G_OBJECT(queue_), "use-buffering", true, nullptr);
}
gst_element_link_many(queue_, audioconvert_, convert_sink, nullptr);
gst_element_link(probe_converter, probe_sink);
// Link the outputs of tee to the queues on each path.
gst_pad_link(gst_element_get_request_pad(tee, "src_%u"),
gst_element_get_static_pad(probe_queue, "sink"));
gst_pad_link(gst_element_get_request_pad(tee, "src_%u"),
gst_element_get_static_pad(audio_queue, "sink"));
// Link replaygain elements if enabled.
if (rg_enabled_) {
gst_element_link_many(rgvolume_, rglimiter_, audioconvert2_, tee, nullptr);
}
// Link the analyzer output of the tee and force 16 bit caps
GstCaps* caps16 = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING,
"S16LE", NULL);
gst_element_link_filtered(probe_queue, probe_converter, caps16);
gst_caps_unref(caps16);
gst_element_link_many(audio_queue, equalizer_preamp_, equalizer_,
stereo_panorama_, volume_, audioscale_, convert,
nullptr);
// We only limit the media type to raw audio.
// Let the audio output of the tee autonegotiate the bit depth and format.
GstCaps* caps = gst_caps_new_empty_simple("audio/x-raw");
// Add caps for fixed sample rate and mono, but only if requested
if (sample_rate_ != GstEngine::kAutoSampleRate && sample_rate_ > 0) {
gst_caps_set_simple(caps, "rate", G_TYPE_INT, sample_rate_, nullptr);
}
if (mono_playback_) {
gst_caps_set_simple(caps, "channels", G_TYPE_INT, 1, nullptr);
}
gst_element_link_filtered(convert, audiosink_, caps);
gst_caps_unref(caps);
// Add probes and handlers.
gst_pad_add_probe(gst_element_get_static_pad(probe_converter, "src"),
GST_PAD_PROBE_TYPE_BUFFER, HandoffCallback, this, nullptr);
gst_bus_set_sync_handler(gst_pipeline_get_bus(GST_PIPELINE(pipeline_)),
BusCallbackSync, this, nullptr);
bus_cb_id_ = gst_bus_add_watch(gst_pipeline_get_bus(GST_PIPELINE(pipeline_)),
BusCallback, this);
MaybeLinkDecodeToAudio();
return true;
}
