Modified: trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp (248720 => 248721)
--- trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2019-08-15 17:15:15 UTC (rev 248720)
+++ trunk/Source/WebCore/platform/mediastream/gstreamer/GStreamerMediaStreamSource.cpp 2019-08-15 17:17:52 UTC (rev 248721)
@@ -171,11 +171,93 @@
gchar* uri;
- GstElement* audioSrc;
- GstClockTime firstAudioBufferPts;
- GstElement* videoSrc;
- GstClockTime firstFramePts;
+ struct SourceData {
+ GRefPtr<GstElement> m_src;
+ void reset(bool isVideo)
+ {
+ m_firstBufferPts = GST_CLOCK_TIME_NONE;
+ m_isVideo = isVideo;
+ m_src = nullptr;
+ }
+ GstElement* src()
+ {
+ return m_src.get();
+ }
+
+ static void needDataCb(GstElement*, guint, WebKitMediaStreamSrc::SourceData *self)
+ {
+ self->setEnoughData(false);
+ }
+
+ static void enoughDataCb(GstElement*, WebKitMediaStreamSrc::SourceData *self)
+ {
+ self->setEnoughData(true);
+ }
+
+
+ void setSrc(GstElement *src)
+ {
+ m_src = adoptGRef(GST_ELEMENT(g_object_ref_sink(src)));
+ if (GST_IS_APP_SRC(src)) {
+ g_object_set(src, "is-live", true, "format", GST_FORMAT_TIME, "emit-signals", TRUE, "min-percent", 100, nullptr);
+ g_signal_connect(src, "enough-data", G_CALLBACK(enoughDataCb), this);
+ g_signal_connect(src, "need-data", G_CALLBACK(needDataCb), this);
+ }
+ }
+
+ bool isUsed()
+ {
+ return !!m_src;
+ }
+
+ void setEnoughData(bool enough)
+ {
+ m_enoughData = enough;
+ }
+
+ void pushSample(GstSample *sample)
+ {
+ if (!m_src)
+ return;
+
+ bool drop = m_enoughData;
+ auto buffer = gst_sample_get_buffer(sample);
+ auto caps = gst_sample_get_caps(sample);
+ if (!GST_CLOCK_TIME_IS_VALID(m_firstBufferPts)) {
+ m_firstBufferPts = GST_BUFFER_PTS(buffer);
+ auto pad = adoptGRef(gst_element_get_static_pad(m_src.get(), "src"));
+ gst_pad_set_offset(pad.get(), -m_firstBufferPts);
+ }
+
+ if (m_isVideo && drop) {
+ drop = (gst_structure_has_name(gst_caps_get_structure(caps, 0), "video/x-raw")
+ || GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT));
+ }
+
+ if (drop) {
+ m_needsDiscont = true;
+ GST_INFO_OBJECT(m_src.get(), "%s queue full already... not pushing", m_isVideo ? "Video" : "Audio");
+ return;
+ }
+
+ if (m_needsDiscont) {
+ GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_FLAG_DISCONT);
+ m_needsDiscont = false;
+ }
+
+ gst_app_src_push_sample(GST_APP_SRC(m_src.get()), sample);
+ }
+ private:
+ GstClockTime m_firstBufferPts;
+ bool m_enoughData;
+ bool m_needsDiscont;
+ bool m_isVideo;
+ };
+
+ SourceData audioSrc;
+ SourceData videoSrc;
+
std::unique_ptr<WebKitMediaStreamTrackObserver> mediaStreamTrackObserver;
std::unique_ptr<WebKitMediaStreamObserver> mediaStreamObserver;
volatile gint npads;
@@ -274,14 +356,14 @@
{
WebKitMediaStreamSrc* self = WEBKIT_MEDIA_STREAM_SRC(object);
- if (self->audioSrc) {
- gst_bin_remove(GST_BIN(self), self->audioSrc);
- self->audioSrc = nullptr;
+ if (self->audioSrc.isUsed()) {
+ gst_bin_remove(GST_BIN(self), self->audioSrc.src());
+ self->audioSrc.reset(false);
}
- if (self->videoSrc) {
- gst_bin_remove(GST_BIN(self), self->videoSrc);
- self->videoSrc = nullptr;
+ if (self->videoSrc.isUsed()) {
+ gst_bin_remove(GST_BIN(self), self->videoSrc.src());
+ self->videoSrc.reset(true);
}
}
@@ -354,8 +436,8 @@
self->mediaStreamTrackObserver = std::make_unique<WebKitMediaStreamTrackObserver>(self);
self->mediaStreamObserver = std::make_unique<WebKitMediaStreamObserver>(self);
self->flowCombiner = gst_flow_combiner_new();
- self->firstAudioBufferPts = GST_CLOCK_TIME_NONE;
- self->firstFramePts = GST_CLOCK_TIME_NONE;
+ self->videoSrc.reset(true);
+ self->audioSrc.reset(false);
}
typedef struct {
@@ -467,13 +549,14 @@
}
static gboolean webkitMediaStreamSrcSetupAppSrc(WebKitMediaStreamSrc* self,
- MediaStreamTrackPrivate* track, GstElement** element,
+ MediaStreamTrackPrivate* track, WebKitMediaStreamSrc::SourceData* data,
GstStaticPadTemplate* pad_template, bool onlyTrack)
{
- *element = gst_element_factory_make("appsrc", nullptr);
- g_object_set(*element, "is-live", true, "format", GST_FORMAT_TIME, nullptr);
+ data->setSrc(gst_element_factory_make("appsrc", nullptr));
+ if (track->isCaptureTrack())
+ g_object_set(data->src(), "do-timestamp", true, nullptr);
- return webkitMediaStreamSrcSetupSrc(self, track, *element, pad_template, TRUE, onlyTrack);
+ return webkitMediaStreamSrcSetupSrc(self, track, data->src(), pad_template, TRUE, onlyTrack);
}
static void webkitMediaStreamSrcPostStreamCollection(WebKitMediaStreamSrc* self, MediaStreamPrivate* stream)
@@ -510,16 +593,16 @@
static void webkitMediaStreamSrcRemoveTrackByType(WebKitMediaStreamSrc* self, RealtimeMediaSource::Type trackType)
{
if (trackType == RealtimeMediaSource::Type::Audio) {
- if (self->audioSrc) {
- gst_element_set_state(self->audioSrc, GST_STATE_NULL);
- gst_bin_remove(GST_BIN(self), self->audioSrc);
- self->audioSrc = nullptr;
+ if (self->audioSrc.isUsed()) {
+ gst_element_set_state(self->audioSrc.src(), GST_STATE_NULL);
+ gst_bin_remove(GST_BIN(self), self->audioSrc.src());
+ self->audioSrc.reset(false);
}
} else if (trackType == RealtimeMediaSource::Type::Video) {
- if (self->videoSrc) {
- gst_element_set_state(self->videoSrc, GST_STATE_NULL);
- gst_bin_remove(GST_BIN(self), self->videoSrc);
- self->videoSrc = nullptr;
+ if (self->videoSrc.isUsed()) {
+ gst_element_set_state(self->videoSrc.src(), GST_STATE_NULL);
+ gst_bin_remove(GST_BIN(self), self->videoSrc.src());
+ self->videoSrc.reset(true);
}
} else
GST_INFO("Unsupported track type: %d", static_cast<int>(trackType));
@@ -544,31 +627,12 @@
static void webkitMediaStreamSrcPushVideoSample(WebKitMediaStreamSrc* self, GstSample* gstsample)
{
- if (self->videoSrc) {
- if (!GST_CLOCK_TIME_IS_VALID(self->firstFramePts)) {
- auto buffer = gst_sample_get_buffer(gstsample);
-
- self->firstFramePts = GST_BUFFER_PTS(buffer);
- auto pad = adoptGRef(gst_element_get_static_pad(self->videoSrc, "src"));
- gst_pad_set_offset(pad.get(), -self->firstFramePts);
- }
-
- gst_app_src_push_sample(GST_APP_SRC(self->videoSrc), gstsample);
- }
+ self->videoSrc.pushSample(gstsample);
}
static void webkitMediaStreamSrcPushAudioSample(WebKitMediaStreamSrc* self, GstSample* gstsample)
{
- if (self->audioSrc) {
- if (!GST_CLOCK_TIME_IS_VALID(self->firstAudioBufferPts)) {
- auto buffer = gst_sample_get_buffer(gstsample);
-
- self->firstAudioBufferPts = GST_BUFFER_PTS(buffer);
- auto pad = adoptGRef(gst_element_get_static_pad(self->audioSrc, "src"));
- gst_pad_set_offset(pad.get(), -self->firstAudioBufferPts);
- }
- gst_app_src_push_sample(GST_APP_SRC(self->audioSrc), gstsample);
- }
+ self->audioSrc.pushSample(gstsample);
}
static void webkitMediaStreamSrcTrackEnded(WebKitMediaStreamSrc* self,
