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<dt>Revision</dt> <dd><a href="http://trac.webkit.org/projects/webkit/changeset/160216">160216</a></dd>
<dt>Author</dt> <dd>philn@webkit.org</dd>
<dt>Date</dt> <dd>2013-12-06 03:20:15 -0800 (Fri, 06 Dec 2013)</dd>
</dl>
<h3>Log Message</h3>
<pre>[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.</pre>
<h3>Modified Paths</h3>
<ul>
<li><a href="#trunkSourceWebCoreChangeLog">trunk/Source/WebCore/ChangeLog</a></li>
<li><a href="#trunkSourceWebCoreplatformaudiogstreamerWebKitWebAudioSourceGStreamercpp">trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp</a></li>
</ul>
</div>
<div id="patch">
<h3>Diff</h3>
<a id="trunkSourceWebCoreChangeLog"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/ChangeLog (160215 => 160216)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/ChangeLog 2013-12-06 11:08:03 UTC (rev 160215)
+++ trunk/Source/WebCore/ChangeLog 2013-12-06 11:20:15 UTC (rev 160216)
</span><span class="lines">@@ -1,3 +1,22 @@
</span><ins>+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.
+
</ins><span class="cx"> 2013-12-05 Philippe Normand <pnormand@igalia.com>
</span><span class="cx">
</span><span class="cx"> [GStreamer] Audio/Video sink management is incoherent
</span></span></pre></div>
<a id="trunkSourceWebCoreplatformaudiogstreamerWebKitWebAudioSourceGStreamercpp"></a>
<div class="modfile"><h4>Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (160215 => 160216)</h4>
<pre class="diff"><span>
<span class="info">--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2013-12-06 11:08:03 UTC (rev 160215)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2013-12-06 11:20:15 UTC (rev 160216)
</span><span class="lines">@@ -60,6 +60,9 @@
</span><span class="cx">
</span><span class="cx"> GSList* pads; // List of queue sink pads. One queue for each planar audio channel.
</span><span class="cx"> GstPad* sourcePad; // src pad of the element, interleaved wav data is pushed to it.
</span><ins>+
+ bool newStreamEventPending;
+ GstSegment segment;
</ins><span class="cx"> };
</span><span class="cx">
</span><span class="cx"> enum {
</span><span class="lines">@@ -181,6 +184,9 @@
</span><span class="cx"> priv->provider = 0;
</span><span class="cx"> priv->bus = 0;
</span><span class="cx">
</span><ins>+ priv->newStreamEventPending = true;
+ gst_segment_init(&priv->segment, GST_FORMAT_TIME);
+
</ins><span class="cx"> g_rec_mutex_init(&priv->mutex);
</span><span class="cx"> priv->task = gst_task_new(reinterpret_cast<GstTaskFunction>(webKitWebAudioSrcLoop), src, 0);
</span><span class="cx">
</span><span class="lines">@@ -215,7 +221,8 @@
</span><span class="cx"> // For each channel of the bus create a new upstream branch for interleave, like:
</span><span class="cx"> // queue ! capsfilter ! audioconvert. which is plugged to a new interleave request sinkpad.
</span><span class="cx"> for (unsigned channelIndex = 0; channelIndex < priv->bus->numberOfChannels(); channelIndex++) {
</span><del>- GstElement* queue = gst_element_factory_make("queue", 0);
</del><ins>+ GOwnPtr<gchar> queueName(g_strdup_printf("webaudioQueue%u", channelIndex));
+ GstElement* queue = gst_element_factory_make("queue", queueName.get());
</ins><span class="cx"> GstElement* capsfilter = gst_element_factory_make("capsfilter", 0);
</span><span class="cx"> GstElement* audioconvert = gst_element_factory_make("audioconvert", 0);
</span><span class="cx">
</span><span class="lines">@@ -334,15 +341,46 @@
</span><span class="cx">
</span><span class="cx"> GSList* padsIt = priv->pads;
</span><span class="cx"> GSList* buffersIt = channelBufferList;
</span><ins>+
+#if GST_CHECK_VERSION(1, 2, 0)
+ guint groupId = 0;
+ if (priv->newStreamEventPending)
+ groupId = gst_util_group_id_next();
+#endif
+
</ins><span class="cx"> for (i = 0; padsIt && buffersIt; padsIt = g_slist_next(padsIt), buffersIt = g_slist_next(buffersIt), ++i) {
</span><span class="cx"> GstPad* pad = static_cast<GstPad*>(padsIt->data);
</span><span class="cx"> GstBuffer* channelBuffer = static_cast<GstBuffer*>(buffersIt->data);
</span><span class="cx">
</span><ins>+ // 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));
+ }
+
</ins><span class="cx"> GstFlowReturn ret = gst_pad_chain(pad, channelBuffer);
</span><span class="cx"> if (ret != GST_FLOW_OK)
</span><del>- GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s:%s", GST_DEBUG_PAD_NAME(pad)));
</del><ins>+ 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)));
</ins><span class="cx"> }
</span><span class="cx">
</span><ins>+ priv->newStreamEventPending = false;
+
</ins><span class="cx"> g_slist_free(channelBufferList);
</span><span class="cx"> }
</span><span class="cx">
</span><span class="lines">@@ -381,6 +419,7 @@
</span><span class="cx"> returnValue = GST_STATE_CHANGE_FAILURE;
</span><span class="cx"> break;
</span><span class="cx"> case GST_STATE_CHANGE_PAUSED_TO_READY:
</span><ins>+ src->priv->newStreamEventPending = true;
</ins><span class="cx"> GST_DEBUG_OBJECT(src, "PAUSED->READY");
</span><span class="cx"> if (!gst_task_join(src->priv->task.get()))
</span><span class="cx"> returnValue = GST_STATE_CHANGE_FAILURE;
</span></span></pre>
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