Added: trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp (0 => 177058)
--- trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp (rev 0)
+++ trunk/Source/WebCore/platform/audio/gstreamer/AudioSourceProviderGStreamer.cpp 2014-12-10 16:00:45 UTC (rev 177058)
@@ -0,0 +1,349 @@
+/*
+ * Copyright (C) 2014 Igalia S.L
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "config.h"
+#include "AudioSourceProviderGStreamer.h"
+
+#if ENABLE(WEB_AUDIO) && ENABLE(VIDEO) && USE(GSTREAMER)
+
+#include "AudioBus.h"
+#include "AudioSourceProviderClient.h"
+#include <gst/app/gstappsink.h>
+#include <gst/audio/audio.h>
+#include <gst/base/gstadapter.h>
+#include <wtf/gobject/GMutexLocker.h>
+
+
+namespace WebCore {
+
+// For now the provider supports only stereo files at a fixed sample
+// bitrate.
+static const int gNumberOfChannels = 2;
+static const float gSampleBitRate = 44100;
+
+static GstFlowReturn onAppsinkNewBufferCallback(GstAppSink* sink, gpointer userData)
+{
+ return static_cast<AudioSourceProviderGStreamer*>(userData)->handleAudioBuffer(sink);
+}
+
+static void onGStreamerDeinterleavePadAddedCallback(GstElement*, GstPad* pad, AudioSourceProviderGStreamer* provider)
+{
+ provider->handleNewDeinterleavePad(pad);
+}
+
+static void onGStreamerDeinterleaveReadyCallback(GstElement*, AudioSourceProviderGStreamer* provider)
+{
+ provider->deinterleavePadsConfigured();
+}
+
+static void onGStreamerDeinterleavePadRemovedCallback(GstElement*, GstPad* pad, AudioSourceProviderGStreamer* provider)
+{
+ provider->handleRemovedDeinterleavePad(pad);
+}
+
+static GstPadProbeReturn onAppsinkFlushCallback(GstPad*, GstPadProbeInfo* info, gpointer userData)
+{
+ if (GST_PAD_PROBE_INFO_TYPE(info) & (GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM | GST_PAD_PROBE_TYPE_EVENT_FLUSH)) {
+ GstEvent* event = GST_PAD_PROBE_INFO_EVENT(info);
+ if (GST_EVENT_TYPE(event) == GST_EVENT_FLUSH_STOP) {
+ AudioSourceProviderGStreamer* provider = reinterpret_cast<AudioSourceProviderGStreamer*>(userData);
+ provider->clearAdapters();
+ }
+ }
+ return GST_PAD_PROBE_OK;
+}
+
+static void copyGStreamerBuffersToAudioChannel(GstAdapter* adapter, AudioBus* bus , int channelNumber, size_t framesToProcess)
+{
+ if (!gst_adapter_available(adapter)) {
+ bus->zero();
+ return;
+ }
+
+ size_t bytes = framesToProcess * sizeof(float);
+ if (gst_adapter_available(adapter) >= bytes) {
+ gst_adapter_copy(adapter, bus->channel(channelNumber)->mutableData(), 0, bytes);
+ gst_adapter_flush(adapter, bytes);
+ }
+}
+
+AudioSourceProviderGStreamer::AudioSourceProviderGStreamer()
+ : m_client(0)
+ , m_deinterleaveSourcePads(0)
+ , m_deinterleavePadAddedHandlerId(0)
+ , m_deinterleaveNoMorePadsHandlerId(0)
+ , m_deinterleavePadRemovedHandlerId(0)
+{
+ g_mutex_init(&m_adapterMutex);
+ m_frontLeftAdapter = gst_adapter_new();
+ m_frontRightAdapter = gst_adapter_new();
+}
+
+AudioSourceProviderGStreamer::~AudioSourceProviderGStreamer()
+{
+ GRefPtr<GstElement> deinterleave = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "deinterleave"));
+ if (deinterleave) {
+ g_signal_handler_disconnect(deinterleave.get(), m_deinterleavePadAddedHandlerId);
+ g_signal_handler_disconnect(deinterleave.get(), m_deinterleaveNoMorePadsHandlerId);
+ g_signal_handler_disconnect(deinterleave.get(), m_deinterleavePadRemovedHandlerId);
+ }
+
+ g_object_unref(m_frontLeftAdapter);
+ g_object_unref(m_frontRightAdapter);
+ g_mutex_clear(&m_adapterMutex);
+}
+
+void AudioSourceProviderGStreamer::configureAudioBin(GstElement* audioBin, GstElement* teePredecessor)
+{
+ m_audioSinkBin = audioBin;
+
+ GstElement* audioTee = gst_element_factory_make("tee", "audioTee");
+ GstElement* audioQueue = gst_element_factory_make("queue", 0);
+ GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
+ GstElement* audioConvert2 = gst_element_factory_make("audioconvert", 0);
+ GstElement* audioResample = gst_element_factory_make("audioresample", 0);
+ GstElement* audioResample2 = gst_element_factory_make("audioresample", 0);
+ GstElement* volumeElement = gst_element_factory_make("volume", "volume");
+ GstElement* audioSink = gst_element_factory_make("autoaudiosink", 0);
+
+ gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), audioTee, audioQueue, audioConvert, audioResample, volumeElement, audioConvert2, audioResample2, audioSink, nullptr);
+
+ // In cases where the audio-sink needs elements before tee (such
+ // as scaletempo) they need to be linked to tee which in this case
+ // doesn't need a ghost pad. It is assumed that the teePredecessor
+ // chain already configured a ghost pad.
+ if (teePredecessor)
+ gst_element_link_pads_full(teePredecessor, "src", audioTee, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ else {
+ // Add a ghostpad to the bin so it can proxy to tee.
+ GRefPtr<GstPad> audioTeeSinkPad = adoptGRef(gst_element_get_static_pad(audioTee, "sink"));
+ gst_element_add_pad(m_audioSinkBin.get(), gst_ghost_pad_new("sink", audioTeeSinkPad.get()));
+ }
+
+ // Link a new src pad from tee to queue ! audioconvert !
+ // audioresample ! volume ! audioconvert ! audioresample !
+ // autoaudiosink. The audioresample and audioconvert are needed to
+ // ensure the audio sink receives buffers in the correct format.
+ gst_element_link_pads_full(audioTee, "src_%u", audioQueue, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(audioQueue, "src", audioConvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(audioConvert, "src", audioResample, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(audioResample, "src", volumeElement, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(volumeElement, "src", audioConvert2, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(audioConvert2, "src", audioResample2, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(audioResample2, "src", audioSink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+}
+
+void AudioSourceProviderGStreamer::provideInput(AudioBus* bus, size_t framesToProcess)
+{
+ GMutexLocker<GMutex> lock(m_adapterMutex);
+ copyGStreamerBuffersToAudioChannel(m_frontLeftAdapter, bus, 0, framesToProcess);
+ copyGStreamerBuffersToAudioChannel(m_frontRightAdapter, bus, 1, framesToProcess);
+}
+
+GstFlowReturn AudioSourceProviderGStreamer::handleAudioBuffer(GstAppSink* sink)
+{
+ if (!m_client)
+ return GST_FLOW_OK;
+
+ // Pull a buffer from appsink and store it the appropriate buffer
+ // list for the audio channel it represents.
+ GRefPtr<GstSample> sample = adoptGRef(gst_app_sink_pull_sample(sink));
+ if (!sample)
+ return gst_app_sink_is_eos(sink) ? GST_FLOW_EOS : GST_FLOW_ERROR;
+
+ GstBuffer* buffer = gst_sample_get_buffer(sample.get());
+ if (!buffer)
+ return GST_FLOW_ERROR;
+
+ GstCaps* caps = gst_sample_get_caps(sample.get());
+ if (!caps)
+ return GST_FLOW_ERROR;
+
+ GstAudioInfo info;
+ gst_audio_info_from_caps(&info, caps);
+
+ GMutexLocker<GMutex> lock(m_adapterMutex);
+
+ // Check the first audio channel. The buffer is supposed to store
+ // data of a single channel anyway.
+ switch (GST_AUDIO_INFO_POSITION(&info, 0)) {
+ case GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT:
+ case GST_AUDIO_CHANNEL_POSITION_MONO:
+ gst_adapter_push(m_frontLeftAdapter, gst_buffer_ref(buffer));
+ break;
+ case GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT:
+ gst_adapter_push(m_frontRightAdapter, gst_buffer_ref(buffer));
+ break;
+ default:
+ break;
+ }
+
+ return GST_FLOW_OK;
+}
+
+void AudioSourceProviderGStreamer::setClient(AudioSourceProviderClient* client)
+{
+ ASSERT(client);
+ m_client = client;
+
+ // The volume element is used to mute audio playback towards the
+ // autoaudiosink. This is needed to avoid double playback of audio
+ // from our audio sink and from the WebAudio AudioDestination node
+ // supposedly configured already by application side.
+ GRefPtr<GstElement> volumeElement = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "volume"));
+ g_object_set(volumeElement.get(), "mute", TRUE, nullptr);
+
+ // The audioconvert and audioresample elements are needed to
+ // ensure deinterleave and the sinks downstream receive buffers in
+ // the format specified by the capsfilter.
+ GstElement* audioQueue = gst_element_factory_make("queue", 0);
+ GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
+ GstElement* audioResample = gst_element_factory_make("audioresample", 0);
+ GstElement* capsFilter = gst_element_factory_make("capsfilter", 0);
+ GstElement* deInterleave = gst_element_factory_make("deinterleave", "deinterleave");
+
+ g_object_set(deInterleave, "keep-positions", TRUE, nullptr);
+ m_deinterleavePadAddedHandlerId = g_signal_connect(deInterleave, "pad-added", G_CALLBACK(onGStreamerDeinterleavePadAddedCallback), this);
+ m_deinterleaveNoMorePadsHandlerId = g_signal_connect(deInterleave, "no-more-pads", G_CALLBACK(onGStreamerDeinterleaveReadyCallback), this);
+ m_deinterleavePadRemovedHandlerId = g_signal_connect(deInterleave, "pad-removed", G_CALLBACK(onGStreamerDeinterleavePadRemovedCallback), this);
+
+ GstCaps* caps = gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(gSampleBitRate),
+ "channels", G_TYPE_INT, gNumberOfChannels,
+ "format", G_TYPE_STRING, GST_AUDIO_NE(F32),
+ "layout", G_TYPE_STRING, "interleaved", nullptr);
+
+ g_object_set(capsFilter, "caps", caps, nullptr);
+ gst_caps_unref(caps);
+
+ gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), audioQueue, audioConvert, audioResample, capsFilter, deInterleave, nullptr);
+
+ GRefPtr<GstElement> audioTee = adoptGRef(gst_bin_get_by_name(GST_BIN(m_audioSinkBin.get()), "audioTee"));
+
+ // Link a new src pad from tee to queue ! audioconvert !
+ // audioresample ! capsfilter ! deinterleave. Later
+ // on each deinterleaved planar audio channel will be routed to an
+ // appsink for data extraction and processing.
+ gst_element_link_pads_full(audioTee.get(), "src_%u", audioQueue, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(audioQueue, "src", audioConvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(audioConvert, "src", audioResample, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(audioResample, "src", capsFilter, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_link_pads_full(capsFilter, "src", deInterleave, "sink", GST_PAD_LINK_CHECK_NOTHING);
+
+ gst_element_sync_state_with_parent(audioQueue);
+ gst_element_sync_state_with_parent(audioConvert);
+ gst_element_sync_state_with_parent(audioResample);
+ gst_element_sync_state_with_parent(capsFilter);
+ gst_element_sync_state_with_parent(deInterleave);
+}
+
+void AudioSourceProviderGStreamer::handleNewDeinterleavePad(GstPad* pad)
+{
+ m_deinterleaveSourcePads++;
+
+ if (m_deinterleaveSourcePads > 2) {
+ g_warning("The AudioSourceProvider supports only mono and stereo audio. Silencing out this new channel.");
+ GstElement* queue = gst_element_factory_make("queue", 0);
+ GstElement* sink = gst_element_factory_make("fakesink", 0);
+ g_object_set(sink, "async", FALSE, nullptr);
+ gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), queue, sink, nullptr);
+
+ GRefPtr<GstPad> sinkPad = adoptGRef(gst_element_get_static_pad(queue, "sink"));
+ gst_pad_link_full(pad, sinkPad.get(), GST_PAD_LINK_CHECK_NOTHING);
+
+ GQuark quark = g_quark_from_static_string("peer");
+ g_object_set_qdata(G_OBJECT(pad), quark, sinkPad.get());
+ gst_element_link_pads_full(queue, "src", sink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+ gst_element_sync_state_with_parent(queue);
+ gst_element_sync_state_with_parent(sink);
+ return;
+ }
+
+ // A new pad for a planar channel was added in deinterleave. Plug
+ // in an appsink so we can pull the data from each
+ // channel. Pipeline looks like:
+ // ... deinterleave ! queue ! appsink.
+ GstElement* queue = gst_element_factory_make("queue", 0);
+ GstElement* sink = gst_element_factory_make("appsink", 0);
+
+ GstAppSinkCallbacks callbacks;
+ callbacks.eos = 0;
+ callbacks.new_preroll = 0;
+ callbacks.new_sample = onAppsinkNewBufferCallback;
+ gst_app_sink_set_callbacks(GST_APP_SINK(sink), &callbacks, this, 0);
+
+ g_object_set(sink, "async", FALSE, nullptr);
+
+ GRefPtr<GstCaps> caps = adoptGRef(gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT, static_cast<int>(gSampleBitRate),
+ "channels", G_TYPE_INT, 1,
+ "format", G_TYPE_STRING, GST_AUDIO_NE(F32),
+ "layout", G_TYPE_STRING, "interleaved", nullptr));
+
+ gst_app_sink_set_caps(GST_APP_SINK(sink), caps.get());
+
+ gst_bin_add_many(GST_BIN(m_audioSinkBin.get()), queue, sink, nullptr);
+
+ GRefPtr<GstPad> sinkPad = adoptGRef(gst_element_get_static_pad(queue, "sink"));
+ gst_pad_link_full(pad, sinkPad.get(), GST_PAD_LINK_CHECK_NOTHING);
+
+ GQuark quark = g_quark_from_static_string("peer");
+ g_object_set_qdata(G_OBJECT(pad), quark, sinkPad.get());
+
+ gst_element_link_pads_full(queue, "src", sink, "sink", GST_PAD_LINK_CHECK_NOTHING);
+
+ sinkPad = adoptGRef(gst_element_get_static_pad(sink, "sink"));
+ gst_pad_add_probe(sinkPad.get(), GST_PAD_PROBE_TYPE_EVENT_FLUSH, onAppsinkFlushCallback, this, nullptr);
+
+ gst_element_sync_state_with_parent(queue);
+ gst_element_sync_state_with_parent(sink);
+}
+
+void AudioSourceProviderGStreamer::handleRemovedDeinterleavePad(GstPad* pad)
+{
+ m_deinterleaveSourcePads--;
+
+ // Remove the queue ! appsink chain downstream of deinterleave.
+ GQuark quark = g_quark_from_static_string("peer");
+ GstPad* sinkPad = reinterpret_cast<GstPad*>(g_object_get_qdata(G_OBJECT(pad), quark));
+ GRefPtr<GstElement> queue = adoptGRef(gst_pad_get_parent_element(sinkPad));
+ GRefPtr<GstPad> queueSrcPad = adoptGRef(gst_element_get_static_pad(queue.get(), "src"));
+ GRefPtr<GstPad> appsinkSinkPad = adoptGRef(gst_pad_get_peer(queueSrcPad.get()));
+ GRefPtr<GstElement> sink = adoptGRef(gst_pad_get_parent_element(appsinkSinkPad.get()));
+ gst_element_set_state(sink.get(), GST_STATE_NULL);
+ gst_element_set_state(queue.get(), GST_STATE_NULL);
+ gst_element_unlink(queue.get(), sink.get());
+ gst_bin_remove_many(GST_BIN(m_audioSinkBin.get()), queue.get(), sink.get(), nullptr);
+}
+
+void AudioSourceProviderGStreamer::deinterleavePadsConfigured()
+{
+ ASSERT(m_client);
+ ASSERT(m_deinterleaveSourcePads == gNumberOfChannels);
+
+ m_client->setFormat(m_deinterleaveSourcePads, gSampleBitRate);
+}
+
+void AudioSourceProviderGStreamer::clearAdapters()
+{
+ GMutexLocker<GMutex> lock(m_adapterMutex);
+ gst_adapter_clear(m_frontLeftAdapter);
+ gst_adapter_clear(m_frontRightAdapter);
+}
+
+} // WebCore
+
+#endif // ENABLE(WEB_AUDIO) && ENABLE(VIDEO) && USE(GSTREAMER)
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp (177057 => 177058)
--- trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2014-12-10 13:00:05 UTC (rev 177057)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp 2014-12-10 16:00:45 UTC (rev 177058)
@@ -64,6 +64,10 @@
#include "WebKitMediaSourceGStreamer.h"
#endif
+#if ENABLE(WEB_AUDIO)
+#include "AudioSourceProviderGStreamer.h"
+#endif
+
// Max interval in seconds to stay in the READY state on manual
// state change requests.
static const unsigned gReadyStateTimerInterval = 60;
@@ -212,6 +216,9 @@
, m_hasAudio(false)
, m_totalBytes(0)
, m_preservesPitch(false)
+#if ENABLE(WEB_AUDIO)
+ , m_audioSourceProvider(AudioSourceProviderGStreamer::create())
+#endif
, m_requestedState(GST_STATE_VOID_PENDING)
, m_missingPlugins(false)
{
@@ -265,6 +272,10 @@
GRefPtr<GstPad> videoSinkPad = adoptGRef(gst_element_get_static_pad(m_webkitVideoSink.get(), "sink"));
g_signal_handlers_disconnect_by_func(videoSinkPad.get(), reinterpret_cast<gpointer>(mediaPlayerPrivateVideoSinkCapsChangedCallback), this);
}
+
+#if ENABLE(WEB_AUDIO)
+ m_audioSourceProvider.release();
+#endif
}
void MediaPlayerPrivateGStreamer::load(const String& urlString)
@@ -1845,35 +1856,56 @@
m_autoAudioSink = gst_element_factory_make("autoaudiosink", 0);
g_signal_connect(m_autoAudioSink.get(), "child-added", G_CALLBACK(setAudioStreamPropertiesCallback), this);
- // Construct audio sink only if pitch preserving is enabled.
- if (!m_preservesPitch)
- return m_autoAudioSink.get();
+ GstElement* audioSinkBin;
- // On 1.4.2 and newer we use the audio-filter property instead.
- if (webkitGstCheckVersion(1, 4, 2))
+ if (webkitGstCheckVersion(1, 4, 2)) {
+#if ENABLE(WEB_AUDIO)
+ audioSinkBin = gst_bin_new("audio-sink");
+ m_audioSourceProvider->configureAudioBin(audioSinkBin, nullptr);
+ return audioSinkBin;
+#else
return m_autoAudioSink.get();
-
- GstElement* scale = gst_element_factory_make("scaletempo", 0);
- if (!scale) {
- GST_WARNING("Failed to create scaletempo");
- return m_autoAudioSink.get();
+#endif
}
- GstElement* audioSinkBin = gst_bin_new("audio-sink");
- GstElement* convert = gst_element_factory_make("audioconvert", 0);
- GstElement* resample = gst_element_factory_make("audioresample", 0);
+ // Construct audio sink only if pitch preserving is enabled.
+ // If GStreamer 1.4.2 is used the audio-filter playbin property is used instead.
+ if (m_preservesPitch) {
+ GstElement* scale = gst_element_factory_make("scaletempo", nullptr);
+ if (!scale) {
+ GST_WARNING("Failed to create scaletempo");
+ return m_autoAudioSink.get();
+ }
- gst_bin_add_many(GST_BIN(audioSinkBin), scale, convert, resample, m_autoAudioSink.get(), NULL);
+ audioSinkBin = gst_bin_new("audio-sink");
+ gst_bin_add(GST_BIN(audioSinkBin), scale);
+ GRefPtr<GstPad> pad = adoptGRef(gst_element_get_static_pad(scale, "sink"));
+ gst_element_add_pad(audioSinkBin, gst_ghost_pad_new("sink", pad.get()));
- if (!gst_element_link_many(scale, convert, resample, m_autoAudioSink.get(), NULL)) {
- GST_WARNING("Failed to link audio sink elements");
- gst_object_unref(audioSinkBin);
- return m_autoAudioSink.get();
+#if ENABLE(WEB_AUDIO)
+ m_audioSourceProvider->configureAudioBin(audioSinkBin, scale);
+#else
+ GstElement* convert = gst_element_factory_make("audioconvert", nullptr);
+ GstElement* resample = gst_element_factory_make("audioresample", nullptr);
+
+ gst_bin_add_many(GST_BIN(audioSinkBin), convert, resample, m_autoAudioSink.get(), nullptr);
+
+ if (!gst_element_link_many(scale, convert, resample, m_autoAudioSink.get(), nullptr)) {
+ GST_WARNING("Failed to link audio sink elements");
+ gst_object_unref(audioSinkBin);
+ return m_autoAudioSink.get();
+ }
+#endif
+ return audioSinkBin;
}
- GRefPtr<GstPad> pad = adoptGRef(gst_element_get_static_pad(scale, "sink"));
- gst_element_add_pad(audioSinkBin, gst_ghost_pad_new("sink", pad.get()));
+#if ENABLE(WEB_AUDIO)
+ audioSinkBin = gst_bin_new("audio-sink");
+ m_audioSourceProvider->configureAudioBin(audioSinkBin, nullptr);
return audioSinkBin;
+#endif
+ ASSERT_NOT_REACHED();
+ return 0;
}
GstElement* MediaPlayerPrivateGStreamer::audioSink() const
