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strawberry-audio-player-win.../src/engine/gstenginepipeline.cpp

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/*
* Strawberry Music Player
* This file was part of Clementine.
* Copyright 2010, David Sansome <me@davidsansome.com>
* Copyright 2018, Jonas Kvinge <jonas@jkvinge.net>
*
* 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 <http://www.gnu.org/licenses/>.
*
*/
#include "config.h"
#include <stdlib.h>
#include <glib.h>
#include <glib-object.h>
#include <gst/gst.h>
#include <QtGlobal>
#include <QObject>
#include <QCoreApplication>
#include <QMutex>
#include <QByteArray>
#include <QList>
#include <QVariant>
#include <QString>
#include <QUrl>
#include <QTimeLine>
#include <QMetaObject>
#include <QtDebug>
#include "core/concurrentrun.h"
#include "core/logging.h"
#include "core/signalchecker.h"
#include "core/timeconstants.h"
#include "enginebase.h"
#include "gstengine.h"
#include "gstenginepipeline.h"
#include "gstbufferconsumer.h"
#include "gstelementdeleter.h"
const int GstEnginePipeline::kGstStateTimeoutNanosecs = 10000000;
const int GstEnginePipeline::kFaderFudgeMsec = 2000;
const int GstEnginePipeline::kEqBandCount = 10;
const int GstEnginePipeline::kEqBandFrequencies[] = { 60, 170, 310, 600, 1000, 3000, 6000, 12000, 14000, 16000 };
int GstEnginePipeline::sId = 1;
GstElementDeleter *GstEnginePipeline::sElementDeleter = nullptr;
GstEnginePipeline::GstEnginePipeline(GstEngine *engine)
: QObject(nullptr),
engine_(engine),
id_(sId++),
valid_(false),
output_(QString()),
device_(QVariant()),
volume_control_(true),
eq_enabled_(false),
eq_preamp_(0),
stereo_balance_(0.0f),
rg_enabled_(false),
rg_mode_(0),
rg_preamp_(0.0),
rg_compression_(true),
buffer_duration_nanosec_(1 * kNsecPerSec),
buffer_min_fill_(33),
buffering_(false),
segment_start_(0),
segment_start_received_(false),
end_offset_nanosec_(-1),
next_beginning_offset_nanosec_(-1),
next_end_offset_nanosec_(-1),
ignore_next_seek_(false),
ignore_tags_(false),
pipeline_is_initialised_(false),
pipeline_is_connected_(false),
pending_seek_nanosec_(-1),
next_uri_set_(false),
volume_percent_(100),
volume_modifier_(1.0),
use_fudge_timer_(false),
pipeline_(nullptr),
audiobin_(nullptr),
queue_(nullptr),
audioconvert_(nullptr),
audioconvert2_(nullptr),
audioscale_(nullptr),
audiosink_(nullptr),
volume_(nullptr),
audio_panorama_(nullptr),
equalizer_preamp_(nullptr),
equalizer_(nullptr),
rgvolume_(nullptr),
rglimiter_(nullptr)
{
if (!sElementDeleter) {
sElementDeleter = new GstElementDeleter(engine_);
}
for (int i = 0; i < kEqBandCount; ++i) eq_band_gains_ << 0;
}
GstEnginePipeline::~GstEnginePipeline() {
if (pipeline_) {
gst_bus_set_sync_handler(gst_pipeline_get_bus(GST_PIPELINE(pipeline_)), nullptr, nullptr, nullptr);
g_source_remove(bus_cb_id_);
gst_element_set_state(pipeline_, GST_STATE_NULL);
gst_object_unref(GST_OBJECT(pipeline_));
}
}
void GstEnginePipeline::set_output_device(const QString &output, const QVariant &device) {
output_ = output;
device_ = device;
}
void GstEnginePipeline::set_volume_control(bool volume_control) {
volume_control_ = volume_control;
}
void GstEnginePipeline::set_replaygain(bool enabled, int mode, float preamp, bool compression) {
rg_enabled_ = enabled;
rg_mode_ = mode;
rg_preamp_ = preamp;
rg_compression_ = compression;
}
void GstEnginePipeline::set_buffer_duration_nanosec(qint64 buffer_duration_nanosec) {
buffer_duration_nanosec_ = buffer_duration_nanosec;
}
void GstEnginePipeline::set_buffer_min_fill(int percent) {
buffer_min_fill_ = percent;
}
bool GstEnginePipeline::InitDecodeBin(GstElement *decode_bin) {
if (!decode_bin) return false;
pipeline_ = gst_pipeline_new("pipeline");
if (!gst_bin_add(GST_BIN(pipeline_), decode_bin)) return false;
if (!InitAudioBin()) return false;
if (!gst_bin_add(GST_BIN(pipeline_), audiobin_)) return false;
if (!gst_element_link(decode_bin, audiobin_)) return false;
return true;
}
GstElement *GstEnginePipeline::CreateDecodeBinFromString(const char *pipeline) {
GError *error = nullptr;
GstElement *bin = gst_parse_bin_from_description(pipeline, TRUE, &error);
if (error) {
QString message = QString::fromLocal8Bit(error->message);
int domain = error->domain;
int code = error->code;
g_error_free(error);
qLog(Warning) << message;
emit Error(id(), message, domain, code);
return nullptr;
}
else {
return bin;
}
}
bool GstEnginePipeline::InitAudioBin() {
// 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");
if (!audiobin_) return false;
// Create the sink
audiosink_ = engine_->CreateElement(output_, audiobin_);
if (!audiosink_) {
gst_object_unref(GST_OBJECT(audiobin_));
return false;
}
if (device_.isValid() && g_object_class_find_property(G_OBJECT_GET_CLASS(audiosink_), "device")) {
switch (device_.type()) {
case QVariant::String:
if (device_.toString().isEmpty()) break;
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;
case QVariant::LongLong:
g_object_set(G_OBJECT(audiosink_), "device", device_.toLongLong(), nullptr);
break;
case QVariant::Int:
g_object_set(G_OBJECT(audiosink_), "device", device_.toInt(), nullptr);
break;
default:
qLog(Warning) << "Unknown device type" << device_;
break;
}
}
// Create all the other elements
queue_ = engine_->CreateElement("queue2", audiobin_);
audioconvert_ = engine_->CreateElement("audioconvert", audiobin_);
GstElement *tee = engine_->CreateElement("tee", audiobin_);
GstElement *probe_queue = engine_->CreateElement("queue2", audiobin_);
GstElement *probe_converter = engine_->CreateElement("audioconvert", audiobin_);
GstElement *probe_sink = engine_->CreateElement("fakesink", audiobin_);
GstElement *audio_queue = engine_->CreateElement("queue", audiobin_);
audioscale_ = engine_->CreateElement("audioresample", audiobin_);
GstElement *convert = engine_->CreateElement("audioconvert", audiobin_);
if (engine_->volume_control()) {
volume_ = engine_->CreateElement("volume", audiobin_);
}
if (eq_enabled_) {
audio_panorama_ = engine_->CreateElement("audiopanorama", audiobin_, false);
equalizer_preamp_ = engine_->CreateElement("volume", audiobin_, false);
equalizer_ = engine_->CreateElement("equalizer-nbands", audiobin_, false);
}
if (!queue_ || !audioconvert_ || !tee || !probe_queue || !probe_converter || !probe_sink || !audio_queue || !audioscale_ || !convert) {
gst_object_unref(GST_OBJECT(audiobin_));
audiobin_ = nullptr;
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_, false);
rglimiter_ = engine_->CreateElement("rglimiter", audiobin_, false);
audioconvert2_ = engine_->CreateElement("audioconvert", audiobin_, false);
if (rgvolume_ && rglimiter_ && audioconvert2_) {
event_probe = audioconvert2_;
convert_sink = rgvolume_;
// 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.
if (equalizer_) {
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.
if (audio_panorama_) g_object_set(G_OBJECT(audio_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);
}
g_object_set(G_OBJECT(probe_queue), "max-size-buffers", 0, nullptr);
g_object_set(G_OBJECT(probe_queue), "max-size-bytes", 0, nullptr);
g_object_set(G_OBJECT(probe_queue), "max-size-time", 0, nullptr);
g_object_set(G_OBJECT(probe_queue), "low-watermark", 0.0, nullptr);
g_object_set(G_OBJECT(probe_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.
pad = gst_element_get_static_pad(probe_queue, "sink");
gst_pad_link(gst_element_get_request_pad(tee, "src_%u"), pad);
gst_object_unref(pad);
pad = gst_element_get_static_pad(audio_queue, "sink");
gst_pad_link(gst_element_get_request_pad(tee, "src_%u"), pad);
gst_object_unref(pad);
// Link replaygain elements if enabled.
if (rg_enabled_ && rgvolume_ && rglimiter_ && audioconvert2_) {
gst_element_link_many(rgvolume_, rglimiter_, audioconvert2_, tee, nullptr);
}
// Don't force 16 bit.
gst_element_link(probe_queue, probe_converter);
if (eq_enabled_ && equalizer_ && equalizer_preamp_ && audio_panorama_) {
if (volume_)
gst_element_link_many(audio_queue, equalizer_preamp_, equalizer_, audio_panorama_, volume_, audioscale_, convert, nullptr);
else
gst_element_link_many(audio_queue, equalizer_preamp_, equalizer_, audio_panorama_, audioscale_, convert, nullptr);
}
else {
if (volume_) gst_element_link_many(audio_queue, volume_, audioscale_, convert, nullptr);
else gst_element_link_many(audio_queue, audioscale_, convert, nullptr);
}
// Let the audio output of the tee autonegotiate the bit depth and format.
GstCaps *caps = gst_caps_new_empty_simple("audio/x-raw");
gst_element_link_filtered(convert, audiosink_, caps);
gst_caps_unref(caps);
// Add probes and handlers.
pad = gst_element_get_static_pad(probe_converter, "src");
gst_pad_add_probe(pad, GST_PAD_PROBE_TYPE_BUFFER, HandoffCallback, this, nullptr);
gst_object_unref(pad);
GstBus *bus = gst_pipeline_get_bus(GST_PIPELINE(pipeline_));
gst_bus_set_sync_handler(bus, BusCallbackSync, this, nullptr);
bus_cb_id_ = gst_bus_add_watch(bus, BusCallback, this);
gst_object_unref(bus);
return true;
}
bool GstEnginePipeline::InitFromString(const QString &pipeline) {
GstElement *new_bin = CreateDecodeBinFromString(pipeline.toUtf8().constData());
if (!new_bin) return false;
return InitDecodeBin(new_bin);
}
bool GstEnginePipeline::InitFromUrl(const QByteArray &media_url, const QUrl original_url, qint64 end_nanosec) {
media_url_ = media_url;
original_url_ = original_url;
end_offset_nanosec_ = end_nanosec;
pipeline_ = engine_->CreateElement("playbin");
if (!pipeline_) return false;
g_object_set(G_OBJECT(pipeline_), "uri", media_url.constData(), nullptr);
CHECKED_GCONNECT(G_OBJECT(pipeline_), "about-to-finish", &AboutToFinishCallback, this);
CHECKED_GCONNECT(G_OBJECT(pipeline_), "pad-added", &NewPadCallback, this);
CHECKED_GCONNECT(G_OBJECT(pipeline_), "notify::source", &SourceSetupCallback, this);
if (!InitAudioBin()) return false;
// Set playbin's sink to be our costum audio-sink.
g_object_set(GST_OBJECT(pipeline_), "audio-sink", audiobin_, NULL);
pipeline_is_connected_ = true;
return true;
}
gboolean GstEnginePipeline::BusCallback(GstBus*, GstMessage *msg, gpointer self) {
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
qLog(Debug) << instance->id() << "bus message" << GST_MESSAGE_TYPE_NAME(msg);
switch (GST_MESSAGE_TYPE(msg)) {
case GST_MESSAGE_ERROR:
instance->ErrorMessageReceived(msg);
break;
case GST_MESSAGE_TAG:
instance->TagMessageReceived(msg);
break;
case GST_MESSAGE_STATE_CHANGED:
instance->StateChangedMessageReceived(msg);
break;
default:
break;
}
return FALSE;
}
GstBusSyncReply GstEnginePipeline::BusCallbackSync(GstBus *, GstMessage *msg, gpointer self) {
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
qLog(Debug) << instance->id() << "sync bus message" << GST_MESSAGE_TYPE_NAME(msg);
switch (GST_MESSAGE_TYPE(msg)) {
case GST_MESSAGE_EOS:
emit instance->EndOfStreamReached(instance->id(), false);
break;
case GST_MESSAGE_TAG:
instance->TagMessageReceived(msg);
break;
case GST_MESSAGE_ERROR:
instance->ErrorMessageReceived(msg);
break;
case GST_MESSAGE_ELEMENT:
instance->ElementMessageReceived(msg);
break;
case GST_MESSAGE_STATE_CHANGED:
instance->StateChangedMessageReceived(msg);
break;
case GST_MESSAGE_BUFFERING:
instance->BufferingMessageReceived(msg);
break;
case GST_MESSAGE_STREAM_STATUS:
instance->StreamStatusMessageReceived(msg);
break;
case GST_MESSAGE_STREAM_START:
instance->StreamStartMessageReceived();
break;
default:
break;
}
return GST_BUS_PASS;
}
void GstEnginePipeline::StreamStatusMessageReceived(GstMessage *msg) {
GstStreamStatusType type;
GstElement *owner;
gst_message_parse_stream_status(msg, &type, &owner);
if (type == GST_STREAM_STATUS_TYPE_CREATE) {
const GValue *val = gst_message_get_stream_status_object(msg);
if (G_VALUE_TYPE(val) == GST_TYPE_TASK) {
GstTask *task = static_cast<GstTask*>(g_value_get_object(val));
gst_task_set_enter_callback(task, &TaskEnterCallback, this, NULL);
}
}
}
void GstEnginePipeline::StreamStartMessageReceived() {
if (next_uri_set_) {
next_uri_set_ = false;
media_url_ = next_media_url_;
original_url_ = next_original_url_;
end_offset_nanosec_ = next_end_offset_nanosec_;
next_media_url_ = QByteArray();
next_original_url_ = QUrl();
next_beginning_offset_nanosec_ = 0;
next_end_offset_nanosec_ = 0;
emit EndOfStreamReached(id(), true);
}
}
void GstEnginePipeline::TaskEnterCallback(GstTask *, GThread *, gpointer) {
// Bump the priority of the thread only on OS X
#ifdef Q_OS_MACOS
sched_param param;
memset(&param, 0, sizeof(param));
param.sched_priority = 99;
pthread_setschedparam(pthread_self(), SCHED_RR, &param);
#endif
}
void GstEnginePipeline::ElementMessageReceived(GstMessage *msg) {
const GstStructure *structure = gst_message_get_structure(msg);
if (gst_structure_has_name(structure, "redirect")) {
const char *uri = gst_structure_get_string(structure, "new-location");
// Set the redirect URL. In mmssrc redirect messages come during the initial state change to PLAYING, so callers can pick up this URL after the state change has failed.
redirect_url_ = uri;
}
}
void GstEnginePipeline::ErrorMessageReceived(GstMessage *msg) {
GError *error = nullptr;
gchar *debugs = nullptr;
gst_message_parse_error(msg, &error, &debugs);
QString message = QString::fromLocal8Bit(error->message);
QString debugstr = QString::fromLocal8Bit(debugs);
int domain = error->domain;
int code = error->code;
g_error_free(error);
free(debugs);
if (state() == GST_STATE_PLAYING && pipeline_is_initialised_ && next_uri_set_ && (domain == GST_RESOURCE_ERROR || domain == GST_STREAM_ERROR)) {
// A track is still playing and the next uri is not playable. We ignore the error here so it can play until the end.
// But there is no message send to the bus when the current track finishes, we have to add an EOS ourself.
qLog(Debug) << "Ignoring error when loading next track";
GstPad *sinkpad = gst_element_get_static_pad(audiobin_, "sink");
gst_pad_send_event(sinkpad, gst_event_new_eos());
gst_object_unref(sinkpad);
return;
}
if (!redirect_url_.isEmpty() && debugstr.contains("A redirect message was posted on the bus and should have been handled by the application.")) {
// mmssrc posts a message on the bus *and* makes an error message when it wants to do a redirect.
// We handle the message, but now we have to ignore the error too.
return;
}
qLog(Error) << __FUNCTION__ << id() << debugstr;
emit Error(id(), message, domain, code);
}
void GstEnginePipeline::TagMessageReceived(GstMessage *msg) {
GstTagList *taglist = nullptr;
gst_message_parse_tag(msg, &taglist);
Engine::SimpleMetaBundle bundle;
bundle.url = original_url_;
bundle.title = ParseStrTag(taglist, GST_TAG_TITLE);
bundle.artist = ParseStrTag(taglist, GST_TAG_ARTIST);
bundle.comment = ParseStrTag(taglist, GST_TAG_COMMENT);
bundle.album = ParseStrTag(taglist, GST_TAG_ALBUM);
bundle.length = 0;
bundle.year = 0;
bundle.track = 0;
bundle.filetype = Song::FileType_Unknown;
bundle.samplerate = 0;
bundle.bitdepth = 0;
bundle.bitrate = ParseUIntTag(taglist, GST_TAG_BITRATE) / 1000;
bundle.lyrics = ParseStrTag(taglist, GST_TAG_LYRICS);
gst_tag_list_free(taglist);
if (ignore_tags_) return;
if (!bundle.title.isEmpty() || !bundle.artist.isEmpty() || !bundle.comment.isEmpty() || !bundle.album.isEmpty())
emit MetadataFound(id(), bundle);
}
QString GstEnginePipeline::ParseStrTag(GstTagList *list, const char *tag) const {
gchar *data = nullptr;
bool success = gst_tag_list_get_string(list, tag, &data);
QString ret;
if (success && data) {
ret = QString::fromUtf8(data);
g_free(data);
}
return ret.trimmed();
}
guint GstEnginePipeline::ParseUIntTag(GstTagList *list, const char *tag) const {
guint data;
bool success = gst_tag_list_get_uint(list, tag, &data);
guint ret = 0;
if (success && data) ret = data;
return ret;
}
void GstEnginePipeline::StateChangedMessageReceived(GstMessage *msg) {
if (msg->src != GST_OBJECT(pipeline_)) {
// We only care about state changes of the whole pipeline.
return;
}
GstState old_state, new_state, pending;
gst_message_parse_state_changed(msg, &old_state, &new_state, &pending);
if (!pipeline_is_initialised_ && (new_state == GST_STATE_PAUSED || new_state == GST_STATE_PLAYING)) {
pipeline_is_initialised_ = true;
if (pending_seek_nanosec_ != -1 && pipeline_is_connected_) {
QMetaObject::invokeMethod(this, "Seek", Qt::QueuedConnection, Q_ARG(qint64, pending_seek_nanosec_));
}
}
if (pipeline_is_initialised_ && new_state != GST_STATE_PAUSED && new_state != GST_STATE_PLAYING) {
pipeline_is_initialised_ = false;
if (next_uri_set_ && new_state == GST_STATE_READY) {
// Revert uri and go back to PLAY state again
next_uri_set_ = false;
g_object_set(G_OBJECT(pipeline_), "uri", media_url_.constData(), nullptr);
SetState(GST_STATE_PLAYING);
}
}
}
void GstEnginePipeline::BufferingMessageReceived(GstMessage *msg) {
// Only handle buffering messages from the queue2 element in audiobin - not the one that's created automatically by uridecodebin.
if (GST_ELEMENT(GST_MESSAGE_SRC(msg)) != queue_) {
return;
}
int percent = 0;
gst_message_parse_buffering(msg, &percent);
const GstState current_state = state();
if (percent == 0 && current_state == GST_STATE_PLAYING && !buffering_) {
buffering_ = true;
emit BufferingStarted();
SetState(GST_STATE_PAUSED);
}
else if (percent == 100 && buffering_) {
buffering_ = false;
emit BufferingFinished();
SetState(GST_STATE_PLAYING);
}
else if (buffering_) {
emit BufferingProgress(percent);
}
}
void GstEnginePipeline::NewPadCallback(GstElement*, GstPad *pad, gpointer self) {
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
GstPad *const audiopad = gst_element_get_static_pad(instance->audiobin_, "sink");
// Link decodebin's sink pad to audiobin's src pad.
if (GST_PAD_IS_LINKED(audiopad)) {
qLog(Warning) << instance->id() << "audiopad is already linked, unlinking old pad";
gst_pad_unlink(audiopad, GST_PAD_PEER(audiopad));
}
gst_pad_link(pad, audiopad);
gst_object_unref(audiopad);
// Offset the timestamps on all the buffers coming out of the decodebin so they line up exactly with the end of the last buffer from the old decodebin.
// "Running time" is the time since the last flushing seek.
GstClockTime running_time = gst_segment_to_running_time(&instance->last_decodebin_segment_, GST_FORMAT_TIME, instance->last_decodebin_segment_.position);
gst_pad_set_offset(pad, running_time);
// Add a probe to the pad so we can update last_decodebin_segment_.
gst_pad_add_probe(pad, static_cast<GstPadProbeType>(GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM | GST_PAD_PROBE_TYPE_EVENT_FLUSH), DecodebinProbe, instance, nullptr);
instance->pipeline_is_connected_ = true;
if (instance->pending_seek_nanosec_ != -1 && instance->pipeline_is_initialised_) {
QMetaObject::invokeMethod(instance, "Seek", Qt::QueuedConnection, Q_ARG(qint64, instance->pending_seek_nanosec_));
}
}
GstPadProbeReturn GstEnginePipeline::DecodebinProbe(GstPad *pad, GstPadProbeInfo *info, gpointer data) {
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(data);
const GstPadProbeType info_type = GST_PAD_PROBE_INFO_TYPE(info);
if (info_type & GST_PAD_PROBE_TYPE_BUFFER) {
// The decodebin produced a buffer. Record its end time, so we can offset the buffers produced by the next decodebin when transitioning to the next song.
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER(info);
GstClockTime timestamp = GST_BUFFER_TIMESTAMP(buffer);
GstClockTime duration = GST_BUFFER_DURATION(buffer);
if (timestamp == GST_CLOCK_TIME_NONE) {
timestamp = instance->last_decodebin_segment_.position;
}
if (duration != GST_CLOCK_TIME_NONE) {
timestamp += duration;
}
instance->last_decodebin_segment_.position = timestamp;
}
else if (info_type & GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) {
GstEvent *event = GST_PAD_PROBE_INFO_EVENT(info);
GstEventType event_type = GST_EVENT_TYPE(event);
if (event_type == GST_EVENT_SEGMENT) {
// A new segment started, we need to save this to calculate running time offsets later.
gst_event_copy_segment(event, &instance->last_decodebin_segment_);
}
else if (event_type == GST_EVENT_FLUSH_START) {
// A flushing seek resets the running time to 0, so remove any offset we set on this pad before.
gst_pad_set_offset(pad, 0);
}
}
return GST_PAD_PROBE_OK;
}
GstPadProbeReturn GstEnginePipeline::HandoffCallback(GstPad*, GstPadProbeInfo *info, gpointer self) {
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
GstBuffer *buf = gst_pad_probe_info_get_buffer(info);
QList<GstBufferConsumer*> consumers;
{
QMutexLocker l(&instance->buffer_consumers_mutex_);
consumers = instance->buffer_consumers_;
}
for (GstBufferConsumer *consumer : consumers) {
gst_buffer_ref(buf);
consumer->ConsumeBuffer(buf, instance->id());
}
// Calculate the end time of this buffer so we can stop playback if it's after the end time of this song.
if (instance->end_offset_nanosec_ > 0) {
quint64 start_time = GST_BUFFER_TIMESTAMP(buf) - instance->segment_start_;
quint64 duration = GST_BUFFER_DURATION(buf);
quint64 end_time = start_time + duration;
if (end_time > instance->end_offset_nanosec_) {
if (instance->has_next_valid_url() && instance->next_media_url_ == instance->media_url_ && instance->next_beginning_offset_nanosec_ == instance->end_offset_nanosec_) {
// The "next" song is actually the next segment of this file - so cheat and keep on playing, but just tell the Engine we've moved on.
instance->end_offset_nanosec_ = instance->next_end_offset_nanosec_;
instance->next_media_url_ = QByteArray();
instance->next_original_url_ = QUrl();
instance->next_beginning_offset_nanosec_ = 0;
instance->next_end_offset_nanosec_ = 0;
// GstEngine will try to seek to the start of the new section, but we're already there so ignore it.
instance->ignore_next_seek_ = true;
emit instance->EndOfStreamReached(instance->id(), true);
}
else {
// There's no next song
emit instance->EndOfStreamReached(instance->id(), false);
}
}
}
return GST_PAD_PROBE_OK;
}
GstPadProbeReturn GstEnginePipeline::EventHandoffCallback(GstPad*, GstPadProbeInfo *info, gpointer self) {
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
GstEvent *e = gst_pad_probe_info_get_event(info);
qLog(Debug) << instance->id() << "event" << GST_EVENT_TYPE_NAME(e);
switch (GST_EVENT_TYPE(e)) {
case GST_EVENT_SEGMENT:
if (!instance->segment_start_received_) {
// The segment start time is used to calculate the proper offset of data buffers from the start of the stream
const GstSegment *segment = nullptr;
gst_event_parse_segment(e, &segment);
instance->segment_start_ = segment->start;
instance->segment_start_received_ = true;
}
break;
default:
break;
}
return GST_PAD_PROBE_OK;
}
void GstEnginePipeline::AboutToFinishCallback(GstPlayBin *bin, gpointer self) {
GstEnginePipeline* instance = reinterpret_cast<GstEnginePipeline*>(self);
if (instance->has_next_valid_url() && !instance->next_uri_set_) {
// Set the next uri. When the current song ends it will be played automatically and a STREAM_START message is send to the bus.
// When the next uri is not playable an error message is send when the pipeline goes to PLAY (or PAUSE) state or immediately if it is currently in PLAY state.
instance->next_uri_set_ = true;
g_object_set(G_OBJECT(instance->pipeline_), "uri", instance->next_media_url_.constData(), nullptr);
}
}
void GstEnginePipeline::SourceSetupCallback(GstPlayBin *bin, GParamSpec *pspec, gpointer self) {
GstEnginePipeline *instance = reinterpret_cast<GstEnginePipeline*>(self);
GstElement *element;
g_object_get(bin, "source", &element, nullptr);
if (!element) {
return;
}
if (g_object_class_find_property(G_OBJECT_GET_CLASS(element), "device") && !instance->source_device().isEmpty()) {
// Gstreamer is not able to handle device in URL (refering to Gstreamer documentation, this might be added in the future).
// Despite that, for now we include device inside URL: we decompose it during Init and set device here, when this callback is called.
g_object_set(element, "device", instance->source_device().toLocal8Bit().constData(), nullptr);
}
if (g_object_class_find_property(G_OBJECT_GET_CLASS(element), "user-agent")) {
QString user_agent = QString("%1 %2").arg(QCoreApplication::applicationName(), QCoreApplication::applicationVersion());
g_object_set(element, "user-agent", user_agent.toUtf8().constData(), nullptr);
g_object_set(element, "ssl-strict", FALSE, nullptr);
//#ifdef Q_OS_MACOS
//g_object_set(element, "tls-database", instance->engine_->tls_database(), nullptr);
//g_object_set(element, "ssl-use-system-ca-file", false, nullptr);
//g_object_set(element, "ssl-strict", TRUE, nullptr);
//#endif
}
// If the pipeline was buffering we stop that now.
if (instance->buffering_) {
instance->buffering_ = false;
emit instance->BufferingFinished();
instance->SetState(GST_STATE_PLAYING);
}
g_object_unref(element);
}
qint64 GstEnginePipeline::position() const {
if (pipeline_is_initialised_)
gst_element_query_position(pipeline_, GST_FORMAT_TIME, &last_known_position_ns_);
return last_known_position_ns_;
}
qint64 GstEnginePipeline::length() const {
gint64 value = 0;
gst_element_query_duration(pipeline_, GST_FORMAT_TIME, &value);
return value;
}
GstState GstEnginePipeline::state() const {
GstState s, sp;
if (gst_element_get_state(pipeline_, &s, &sp, kGstStateTimeoutNanosecs) == GST_STATE_CHANGE_FAILURE)
return GST_STATE_NULL;
return s;
}
QFuture<GstStateChangeReturn> GstEnginePipeline::SetState(GstState state) {
return ConcurrentRun::Run<GstStateChangeReturn, GstElement*, GstState>(&set_state_threadpool_, &gst_element_set_state, pipeline_, state);
}
bool GstEnginePipeline::Seek(qint64 nanosec) {
if (ignore_next_seek_) {
ignore_next_seek_ = false;
return true;
}
if (!pipeline_is_connected_ || !pipeline_is_initialised_) {
pending_seek_nanosec_ = nanosec;
return true;
}
if (next_uri_set_) {
qDebug() << "MYTODO: gstenginepipeline.seek: seeking after Transition";
pending_seek_nanosec_ = nanosec;
SetState(GST_STATE_READY);
return true;
}
pending_seek_nanosec_ = -1;
last_known_position_ns_ = nanosec;
return gst_element_seek_simple(pipeline_, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, nanosec);
}
void GstEnginePipeline::SetEqualizerEnabled(bool enabled) {
eq_enabled_ = enabled;
UpdateEqualizer();
}
void GstEnginePipeline::SetEqualizerParams(int preamp, const QList<int>& band_gains) {
eq_preamp_ = preamp;
eq_band_gains_ = band_gains;
UpdateEqualizer();
}
void GstEnginePipeline::SetStereoBalance(float value) {
stereo_balance_ = value;
UpdateStereoBalance();
}
void GstEnginePipeline::UpdateEqualizer() {
if (!equalizer_ || !equalizer_preamp_) return;
// Update band gains
for (int i = 0; i < kEqBandCount; ++i) {
float gain = eq_enabled_ ? eq_band_gains_[i] : 0.0;
if (gain < 0)
gain *= 0.24;
else
gain *= 0.12;
const int index_in_eq = i + 1;
// Offset because of the first dummy band we created.
GstObject *band = GST_OBJECT(gst_child_proxy_get_child_by_index(GST_CHILD_PROXY(equalizer_), index_in_eq));
g_object_set(G_OBJECT(band), "gain", gain, nullptr);
g_object_unref(G_OBJECT(band));
}
// Update preamp
float preamp = 1.0;
if (eq_enabled_) preamp = float(eq_preamp_ + 100) * 0.01; // To scale from 0.0 to 2.0
g_object_set(G_OBJECT(equalizer_preamp_), "volume", preamp, nullptr);
}
void GstEnginePipeline::UpdateStereoBalance() {
if (audio_panorama_) {
g_object_set(G_OBJECT(audio_panorama_), "panorama", stereo_balance_, nullptr);
}
}
void GstEnginePipeline::SetVolume(int percent) {
if (!volume_) return;
volume_percent_ = percent;
UpdateVolume();
}
void GstEnginePipeline::SetVolumeModifier(qreal mod) {
if (!volume_) return;
volume_modifier_ = mod;
UpdateVolume();
}
void GstEnginePipeline::UpdateVolume() {
if (!volume_) return;
float vol = double(volume_percent_) * 0.01 * volume_modifier_;
g_object_set(G_OBJECT(volume_), "volume", vol, nullptr);
}
void GstEnginePipeline::StartFader(qint64 duration_nanosec, QTimeLine::Direction direction, QTimeLine::CurveShape shape, bool use_fudge_timer) {
const int duration_msec = duration_nanosec / kNsecPerMsec;
// If there's already another fader running then start from the same time that one was already at.
int start_time = direction == QTimeLine::Forward ? 0 : duration_msec;
if (fader_ && fader_->state() == QTimeLine::Running) {
if (duration_msec == fader_->duration()) {
start_time = fader_->currentTime();
}
else {
// Calculate the position in the new fader with the same value from the old fader, so no volume jumps appear
qreal time = qreal(duration_msec) * (qreal(fader_->currentTime()) / qreal(fader_->duration()));
start_time = qRound(time);
}
}
fader_.reset(new QTimeLine(duration_msec, this));
connect(fader_.get(), SIGNAL(valueChanged(qreal)), SLOT(SetVolumeModifier(qreal)));
connect(fader_.get(), SIGNAL(finished()), SLOT(FaderTimelineFinished()));
fader_->setDirection(direction);
fader_->setCurveShape(shape);
fader_->setCurrentTime(start_time);
fader_->resume();
fader_fudge_timer_.stop();
use_fudge_timer_ = use_fudge_timer;
SetVolumeModifier(fader_->currentValue());
}
void GstEnginePipeline::FaderTimelineFinished() {
fader_.reset();
// Wait a little while longer before emitting the finished signal (and probably distroying the pipeline) to account for delays in the audio server/driver.
if (use_fudge_timer_) {
fader_fudge_timer_.start(kFaderFudgeMsec, this);
}
else {
// Even here we cannot emit the signal directly, as it result in a stutter when resuming playback.
// So use a quest small time, so you won't notice the difference when resuming playback
// (You get here when the pause fading is active)
fader_fudge_timer_.start(250, this);
}
}
void GstEnginePipeline::timerEvent(QTimerEvent *e) {
if (e->timerId() == fader_fudge_timer_.timerId()) {
fader_fudge_timer_.stop();
emit FaderFinished();
return;
}
QObject::timerEvent(e);
}
void GstEnginePipeline::AddBufferConsumer(GstBufferConsumer *consumer) {
QMutexLocker l(&buffer_consumers_mutex_);
buffer_consumers_ << consumer;
}
void GstEnginePipeline::RemoveBufferConsumer(GstBufferConsumer *consumer) {
QMutexLocker l(&buffer_consumers_mutex_);
buffer_consumers_.removeAll(consumer);
}
void GstEnginePipeline::RemoveAllBufferConsumers() {
QMutexLocker l(&buffer_consumers_mutex_);
buffer_consumers_.clear();
}
void GstEnginePipeline::SetNextUrl(const QByteArray &media_url, const QUrl &original_url, qint64 beginning_nanosec, qint64 end_nanosec) {
next_media_url_ = media_url;
next_original_url_ = original_url;
next_beginning_offset_nanosec_ = beginning_nanosec;
next_end_offset_nanosec_ = end_nanosec;
}
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