Changeset 160216 in webkit

Timestamp:

Dec 6, 2013 3:20:15 AM (10 years ago)

Author:

Philippe Normand

Message:

[GStreamer] webkitwebaudiosrc element needs to emit stream-start, caps and segment events
​https://bugs.webkit.org/show_bug.cgi?id=123015

Reviewed by Martin Robinson.

When the source element starts emitting buffers send along various
events to notify downstream elements.

No new tests, change covered by existing webaudio tests.

• platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:

(webkit_web_audio_src_init): Initialize segment.
(webKitWebAudioSrcConstructed): Give an explicit name to each
queue added in front of the interleave element.
(webKitWebAudioSrcLoop): Before sending the first buffers push
stream-start, caps and segment events on each queue's sinkpad.

Location:

trunk/Source/WebCore

Files:

• ChangeLog (modified) (1 diff)
• platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (modified) (5 diffs)

trunk/Source/WebCore/ChangeLog

@@ +1 @@
+2013-11-11  Philippe Normand  <pnormand@igalia.com>
+
+        [GStreamer] webkitwebaudiosrc element needs to emit stream-start, caps and segment events
+        https://bugs.webkit.org/show_bug.cgi?id=123015
+
+        Reviewed by Martin Robinson.
+
+        When the source element starts emitting buffers send along various
+        events to notify downstream elements.
+
+        No new tests, change covered by existing webaudio tests.
+
+        * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+        (webkit_web_audio_src_init): Initialize segment.
+        (webKitWebAudioSrcConstructed): Give an explicit name to each
+        queue added in front of the interleave element.
+        (webKitWebAudioSrcLoop): Before sending the first buffers push
+        stream-start, caps and segment events on each queue's sinkpad.
+
 2013-12-05  Philippe Normand  <pnormand@igalia.com>
 

trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp

@@ -61 +61 @@
     GSList* pads; // List of queue sink pads. One queue for each planar audio channel.
     GstPad* sourcePad; // src pad of the element, interleaved wav data is pushed to it.
+
+    bool newStreamEventPending;
+    GstSegment segment;
 };
 
@@ -182 +185 @@
     priv->bus = 0;
 
+    priv->newStreamEventPending = true;
+    gst_segment_init(&priv->segment, GST_FORMAT_TIME);
+
     g_rec_mutex_init(&priv->mutex);
     priv->task = gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, 0);
@@ -216 +222 @@
     // queue ! capsfilter ! audioconvert. which is plugged to a new interleave request sinkpad.
     for (unsigned channelIndex = 0; channelIndex < priv->bus->numberOfChannels(); channelIndex++) {
-        GstElement* queue = gst_element_factory_make("queue", 0);
+        GOwnPtr<gchar> queueName(g_strdup_printf("webaudioQueue%u", channelIndex));
+        GstElement* queue = gst_element_factory_make("queue", queueName.get());
         GstElement* capsfilter = gst_element_factory_make("capsfilter", 0);
         GstElement* audioconvert = gst_element_factory_make("audioconvert", 0);
@@ -335 +342 @@
     GSList* padsIt = priv->pads;
     GSList* buffersIt = channelBufferList;
+
+#if GST_CHECK_VERSION(1, 2, 0)
+    guint groupId = 0;
+    if (priv->newStreamEventPending)
+        groupId = gst_util_group_id_next();
+#endif
+
     for (i = 0; padsIt && buffersIt; padsIt = g_slist_next(padsIt), buffersIt = g_slist_next(buffersIt), ++i) {
         GstPad* pad = static_cast<GstPad*>(padsIt->data);
         GstBuffer* channelBuffer = static_cast<GstBuffer*>(buffersIt->data);
 
+        // Send stream-start, segment and caps events downstream, along with the first buffer.
+        if (priv->newStreamEventPending) {
+            GRefPtr<GstElement> queue = adoptGRef(gst_pad_get_parent_element(pad));
+            GRefPtr<GstPad> sinkPad = adoptGRef(gst_element_get_static_pad(queue.get(), "sink"));
+            GOwnPtr<gchar> queueName(gst_element_get_name(queue.get()));
+            GOwnPtr<gchar> streamId(g_strdup_printf("webaudio/%s", queueName.get()));
+            GstEvent* streamStartEvent = gst_event_new_stream_start(streamId.get());
+#if GST_CHECK_VERSION(1, 2, 0)
+            gst_event_set_group_id(streamStartEvent, groupId);
+#endif
+            gst_pad_send_event(sinkPad.get(), streamStartEvent);
+
+            GRefPtr<GstCaps> monoCaps = adoptGRef(getGStreamerMonoAudioCaps(priv->sampleRate));
+            GstAudioInfo info;
+            gst_audio_info_from_caps(&info, monoCaps.get());
+            GST_AUDIO_INFO_POSITION(&info, 0) = webKitWebAudioGStreamerChannelPosition(i);
+            GRefPtr<GstCaps> capsWithChannelPosition = adoptGRef(gst_audio_info_to_caps(&info));
+            gst_pad_send_event(sinkPad.get(), gst_event_new_caps(capsWithChannelPosition.get()));
+
+            gst_pad_send_event(sinkPad.get(), gst_event_new_segment(&priv->segment));
+        }
+
         GstFlowReturn ret = gst_pad_chain(pad, channelBuffer);
         if (ret != GST_FLOW_OK)
-            GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s:%s", GST_DEBUG_PAD_NAME(pad)));
-    }
+            GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s:%s flow: %s", GST_DEBUG_PAD_NAME(pad), gst_flow_get_name(ret)));
+    }
+
+    priv->newStreamEventPending = false;
 
     g_slist_free(channelBufferList);
@@ -382 +420 @@
         break;
     case GST_STATE_CHANGE_PAUSED_TO_READY:
+        src->priv->newStreamEventPending = true;
         GST_DEBUG_OBJECT(src, "PAUSED->READY");
         if (!gst_task_join(src->priv->task.get()))