void AudioFileReader::decodeAudioForBusCreation()
{
// Build the pipeline (giostreamsrc | filesrc) ! decodebin2
// A deinterleave element is added once a src pad becomes available in decodebin.
m_pipeline = gst_pipeline_new(0);
GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
ASSERT(bus);
gst_bus_add_signal_watch(bus.get());
g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
GstElement* source;
if (m_data) {
ASSERT(m_dataSize);
source = gst_element_factory_make("giostreamsrc", 0);
GRefPtr<GInputStream> memoryStream = adoptGRef(g_memory_input_stream_new_from_data(m_data, m_dataSize, 0));
g_object_set(source, "stream", memoryStream.get(), NULL);
} else {
source = gst_element_factory_make("filesrc", 0);
g_object_set(source, "location", m_filePath, NULL);
}
m_decodebin = gst_element_factory_make(gDecodebinName, "decodebin");
g_signal_connect(m_decodebin.get(), "pad-added", G_CALLBACK(onGStreamerDecodebinPadAddedCallback), this);
gst_bin_add_many(GST_BIN(m_pipeline), source, m_decodebin.get(), NULL);
gst_element_link_pads_full(source, "src", m_decodebin.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_set_state(m_pipeline, GST_STATE_PAUSED);
}
AudioFileReader::~AudioFileReader()
{
if (m_pipeline) {
GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
ASSERT(bus);
g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
gst_bus_remove_signal_watch(bus.get());
gst_element_set_state(m_pipeline, GST_STATE_NULL);
gst_object_unref(GST_OBJECT(m_pipeline));
}
if (m_decodebin) {
g_signal_handlers_disconnect_by_func(m_decodebin.get(), reinterpret_cast<gpointer>(onGStreamerDecodebinPadAddedCallback), this);
m_decodebin.clear();
}
if (m_deInterleave) {
g_signal_handlers_disconnect_by_func(m_deInterleave.get(), reinterpret_cast<gpointer>(onGStreamerDeinterleavePadAddedCallback), this);
g_signal_handlers_disconnect_by_func(m_deInterleave.get(), reinterpret_cast<gpointer>(onGStreamerDeinterleaveReadyCallback), this);
m_deInterleave.clear();
}
#ifndef GST_API_VERSION_1
gst_buffer_list_iterator_free(m_frontLeftBuffersIterator);
gst_buffer_list_iterator_free(m_frontRightBuffersIterator);
#endif
gst_buffer_list_unref(m_frontLeftBuffers);
gst_buffer_list_unref(m_frontRightBuffers);
}
AudioDestinationGStreamer::~AudioDestinationGStreamer()
{
GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
ASSERT(bus);
g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(messageCallback), this);
gst_bus_remove_signal_watch(bus.get());
gst_element_set_state(m_pipeline, GST_STATE_NULL);
gst_object_unref(m_pipeline);
}
GStreamerGWorld::GStreamerGWorld(GstElement* pipeline)
: m_pipeline(pipeline)
{
// XOverlay messages need to be handled synchronously.
GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
#ifndef GST_API_VERSION_1
gst_bus_set_sync_handler(bus.get(), gst_bus_sync_signal_handler, this);
#else
gst_bus_set_sync_handler(bus.get(), gst_bus_sync_signal_handler, this, 0);
#endif
g_signal_connect(bus.get(), "sync-message::element", G_CALLBACK(gstGWorldSyncMessageCallback), this);
}
GStreamerGWorld::~GStreamerGWorld()
{
exitFullscreen();
GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
g_signal_handlers_disconnect_by_func(bus.get(), reinterpret_cast<gpointer>(gstGWorldSyncMessageCallback), this);
#ifndef GST_API_VERSION_1
gst_bus_set_sync_handler(bus.get(), 0, this);
#else
gst_bus_set_sync_handler(bus.get(), 0, this, 0);
#endif
m_pipeline = 0;
}
AudioDestinationGStreamer::AudioDestinationGStreamer(AudioIOCallback& callback, float sampleRate)
: m_callback(callback)
, m_renderBus(2, framesToPull, false)
, m_sampleRate(sampleRate)
, m_isPlaying(false)
{
m_pipeline = gst_pipeline_new("play");
GRefPtr<GstBus> bus = webkitGstPipelineGetBus(GST_PIPELINE(m_pipeline));
ASSERT(bus);
gst_bus_add_signal_watch(bus.get());
g_signal_connect(bus.get(), "message", G_CALLBACK(messageCallback), this);
GstElement* webkitAudioSrc = reinterpret_cast<GstElement*>(g_object_new(WEBKIT_TYPE_WEB_AUDIO_SRC,
"rate", sampleRate,
"bus", &m_renderBus,
"provider", &m_callback,
"frames", framesToPull, NULL));
GstElement* wavParser = gst_element_factory_make("wavparse", 0);
m_wavParserAvailable = wavParser;
ASSERT_WITH_MESSAGE(m_wavParserAvailable, "Failed to create GStreamer wavparse element");
if (!m_wavParserAvailable)
return;
#ifndef GST_API_VERSION_1
g_signal_connect(wavParser, "pad-added", G_CALLBACK(onGStreamerWavparsePadAddedCallback), this);
#endif
gst_bin_add_many(GST_BIN(m_pipeline), webkitAudioSrc, wavParser, NULL);
gst_element_link_pads_full(webkitAudioSrc, "src", wavParser, "sink", GST_PAD_LINK_CHECK_NOTHING);
#ifdef GST_API_VERSION_1
GRefPtr<GstPad> srcPad = adoptGRef(gst_element_get_static_pad(wavParser, "src"));
finishBuildingPipelineAfterWavParserPadReady(srcPad.get());
#endif
}