Modified: trunk/Source/WebCore/ChangeLog (177078 => 177079)
--- trunk/Source/WebCore/ChangeLog 2014-12-10 18:37:39 UTC (rev 177078)
+++ trunk/Source/WebCore/ChangeLog 2014-12-10 18:38:25 UTC (rev 177079)
@@ -1,3 +1,16 @@
+2014-12-10 Sebastian Dröge <[email protected]>
+
+ [GStreamer] Use a buffer pool for allocations in the AudioDestination
+ https://bugs.webkit.org/show_bug.cgi?id=139487
+
+ Reviewed by Philippe Normand.
+
+ * platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp:
+ (webKitWebAudioSrcLoop):
+ (webKitWebAudioSrcChangeState):
+ Use a buffer pool for the allocations instead of allocating a
+ new small buffer for every 128 samples we produce.
+
2014-12-10 Martin Robinson <[email protected]>
[GTK] Add support for text-decoration-skip
Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (177078 => 177079)
--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2014-12-10 18:37:39 UTC (rev 177078)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2014-12-10 18:38:25 UTC (rev 177079)
@@ -64,6 +64,8 @@
bool newStreamEventPending;
GstSegment segment;
guint64 numberOfSamples;
+
+ GstBufferPool* pool;
};
enum {
@@ -318,6 +320,7 @@
ASSERT(priv->bus);
ASSERT(priv->provider);
if (!priv->provider || !priv->bus) {
+ GST_ELEMENT_ERROR(src, CORE, FAILED, ("Internal WebAudioSrc error"), ("Can't start without provider or bus"));
gst_task_pause(src->priv->task.get());
return;
}
@@ -328,10 +331,26 @@
GSList* channelBufferList = 0;
register int i;
- unsigned bufferSize = priv->framesToPull * sizeof(float);
for (i = g_slist_length(priv->pads) - 1; i >= 0; i--) {
AudioSrcBuffer* buffer = g_new(AudioSrcBuffer, 1);
- GstBuffer* channelBuffer = gst_buffer_new_and_alloc(bufferSize);
+ GstBuffer* channelBuffer;
+
+ GstFlowReturn ret = gst_buffer_pool_acquire_buffer(priv->pool, &channelBuffer, nullptr);
+
+ if (ret != GST_FLOW_OK) {
+ g_free(buffer);
+ while (channelBufferList) {
+ buffer = static_cast<AudioSrcBuffer*>(channelBufferList->data);
+ gst_buffer_unmap(buffer->buffer, &buffer->info);
+ gst_buffer_unref(buffer->buffer);
+ g_free(buffer);
+ channelBufferList = g_slist_delete_link(channelBufferList, channelBufferList);
+ }
+ GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to allocate buffer for flow: %s", gst_flow_get_name(ret)));
+ gst_task_pause(src->priv->task.get());
+ return;
+ }
+
ASSERT(channelBuffer);
buffer->buffer = channelBuffer;
GST_BUFFER_TIMESTAMP(channelBuffer) = timestamp;
@@ -421,16 +440,26 @@
}
switch (transition) {
- case GST_STATE_CHANGE_READY_TO_PAUSED:
+ case GST_STATE_CHANGE_READY_TO_PAUSED: {
GST_DEBUG_OBJECT(src, "READY->PAUSED");
- if (!gst_task_start(src->priv->task.get()))
+ src->priv->pool = gst_buffer_pool_new();
+ GstStructure* config = gst_buffer_pool_get_config(src->priv->pool);
+ gst_buffer_pool_config_set_params(config, nullptr, src->priv->framesToPull * sizeof(float), 0, 0);
+ gst_buffer_pool_set_config(src->priv->pool, config);
+ if (!gst_buffer_pool_set_active(src->priv->pool, TRUE))
returnValue = GST_STATE_CHANGE_FAILURE;
+ else if (!gst_task_start(src->priv->task.get()))
+ returnValue = GST_STATE_CHANGE_FAILURE;
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()))
returnValue = GST_STATE_CHANGE_FAILURE;
+ gst_buffer_pool_set_active(src->priv->pool, FALSE);
+ gst_object_unref(src->priv->pool);
+ src->priv->pool = nullptr;
break;
default:
break;
