Modified: trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp (210583 => 210584)
--- trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2017-01-11 09:08:11 UTC (rev 210583)
+++ trunk/Source/WebCore/platform/audio/gstreamer/WebKitWebAudioSourceGStreamer.cpp 2017-01-11 11:27:38 UTC (rev 210584)
@@ -60,12 +60,15 @@
GRefPtr<GstTask> task;
GRecMutex mutex;
- GSList* sources; // List of appsrc. One appsrc for each planar audio channel.
- GstPad* sourcePad; // src pad of the element, interleaved wav data is pushed to it.
+ // List of appsrc. One appsrc for each planar audio channel.
+ Vector<GRefPtr<GstElement>> sources;
+ // src pad of the element, interleaved wav data is pushed to it.
+ GstPad* sourcePad;
+
guint64 numberOfSamples;
- GstBufferPool* pool;
+ GRefPtr<GstBufferPool> pool;
};
enum {
@@ -75,11 +78,6 @@
PROP_FRAMES
};
-typedef struct {
- GstBuffer* buffer;
- GstMapInfo info;
-} AudioSrcBuffer;
-
static GstStaticPadTemplate srcTemplate = GST_STATIC_PAD_TEMPLATE("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
@@ -220,7 +218,7 @@
// appsrc ! . which is plugged to a new interleave request sinkpad.
for (unsigned channelIndex = 0; channelIndex < priv->bus->numberOfChannels(); channelIndex++) {
GUniquePtr<gchar> appsrcName(g_strdup_printf("webaudioSrc%u", channelIndex));
- GstElement* appsrc = gst_element_factory_make("appsrc", appsrcName.get());
+ GRefPtr<GstElement> appsrc = gst_element_factory_make("appsrc", appsrcName.get());
GRefPtr<GstCaps> monoCaps = adoptGRef(getGStreamerMonoAudioCaps(priv->sampleRate));
GstAudioInfo info;
@@ -229,16 +227,15 @@
GRefPtr<GstCaps> caps = adoptGRef(gst_audio_info_to_caps(&info));
// Configure the appsrc for minimal latency.
- g_object_set(appsrc, "max-bytes", static_cast<guint64>(2 * priv->bufferSize), "block", TRUE,
+ g_object_set(appsrc.get(), "max-bytes", static_cast<guint64>(2 * priv->bufferSize), "block", TRUE,
"blocksize", priv->bufferSize,
"format", GST_FORMAT_TIME, "caps", caps.get(), nullptr);
- priv->sources = g_slist_prepend(priv->sources, gst_object_ref(appsrc));
+ priv->sources.append(appsrc);
- gst_bin_add(GST_BIN(src), appsrc);
- gst_element_link_pads_full(appsrc, "src", priv->interleave.get(), "sink_%u", GST_PAD_LINK_CHECK_NOTHING);
+ gst_bin_add(GST_BIN(src), appsrc.get());
+ gst_element_link_pads_full(appsrc.get(), "src", priv->interleave.get(), "sink_%u", GST_PAD_LINK_CHECK_NOTHING);
}
- priv->sources = g_slist_reverse(priv->sources);
// interleave's src pad is the only visible pad of our element.
GRefPtr<GstPad> targetPad = adoptGRef(gst_element_get_static_pad(priv->interleave.get(), "src"));
@@ -252,8 +249,6 @@
g_rec_mutex_clear(&priv->mutex);
- g_slist_free_full(priv->sources, reinterpret_cast<GDestroyNotify>(gst_object_unref));
-
priv->~WebKitWebAudioSourcePrivate();
GST_CALL_PARENT(G_OBJECT_CLASS, finalize, ((GObject* )(src)));
}
@@ -319,26 +314,19 @@
return;
}
+ ASSERT(priv->pool);
GstClockTime timestamp = gst_util_uint64_scale(priv->numberOfSamples, GST_SECOND, priv->sampleRate);
priv->numberOfSamples += priv->framesToPull;
GstClockTime duration = gst_util_uint64_scale(priv->numberOfSamples, GST_SECOND, priv->sampleRate) - timestamp;
- GSList* channelBufferList = 0;
- for (int i = g_slist_length(priv->sources) - 1; i >= 0; i--) {
- AudioSrcBuffer* buffer = g_new(AudioSrcBuffer, 1);
- GstBuffer* channelBuffer;
-
- GstFlowReturn ret = gst_buffer_pool_acquire_buffer(priv->pool, &channelBuffer, nullptr);
-
+ Vector<GRefPtr<GstBuffer>> channelBufferList;
+ channelBufferList.reserveInitialCapacity(priv->sources.size());
+ for (unsigned i = 0; i < priv->sources.size(); ++i) {
+ GRefPtr<GstBuffer> buffer;
+ GstFlowReturn ret = gst_buffer_pool_acquire_buffer(priv->pool.get(), &buffer.outPtr(), 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);
- }
+ for (auto& buffer : channelBufferList)
+ unmapGstBuffer(buffer.get());
// FLUSHING and EOS are not errors.
if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
@@ -347,44 +335,38 @@
return;
}
- ASSERT(channelBuffer);
- buffer->buffer = channelBuffer;
- GST_BUFFER_TIMESTAMP(channelBuffer) = timestamp;
- GST_BUFFER_DURATION(channelBuffer) = duration;
- gst_buffer_map(channelBuffer, &buffer->info, (GstMapFlags) GST_MAP_READWRITE);
- priv->bus->setChannelMemory(i, reinterpret_cast<float*>(buffer->info.data), priv->framesToPull);
- channelBufferList = g_slist_prepend(channelBufferList, buffer);
+ ASSERT(buffer);
+ GST_BUFFER_TIMESTAMP(buffer.get()) = timestamp;
+ GST_BUFFER_DURATION(buffer.get()) = duration;
+ mapGstBuffer(buffer.get(), GST_MAP_READWRITE);
+ priv->bus->setChannelMemory(i, reinterpret_cast<float*>(getGstBufferDataPointer(buffer.get())), priv->framesToPull);
+ channelBufferList.uncheckedAppend(WTFMove(buffer));
}
// FIXME: Add support for local/live audio input.
priv->provider->render(0, priv->bus, priv->framesToPull);
- GSList* sourcesIt = priv->sources;
- GSList* buffersIt = channelBufferList;
+ ASSERT(channelBufferList.size() == priv->sources.size());
+ bool failed = false;
+ for (unsigned i = 0; i < priv->sources.size(); ++i) {
+ // Unmap before passing on the buffer.
+ auto& buffer = channelBufferList[i];
+ unmapGstBuffer(buffer.get());
- GstFlowReturn ret = GST_FLOW_OK;
- for (int i = 0; sourcesIt && buffersIt; sourcesIt = g_slist_next(sourcesIt), buffersIt = g_slist_next(buffersIt), ++i) {
- GstElement* appsrc = static_cast<GstElement*>(sourcesIt->data);
- AudioSrcBuffer* buffer = static_cast<AudioSrcBuffer*>(buffersIt->data);
- GstBuffer* channelBuffer = buffer->buffer;
+ if (failed)
+ continue;
- // Unmap before passing on the buffer.
- gst_buffer_unmap(channelBuffer, &buffer->info);
- g_free(buffer);
-
- if (ret == GST_FLOW_OK) {
- ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc), channelBuffer);
- if (ret != GST_FLOW_OK) {
- // FLUSHING and EOS are not errors.
- if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
- GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s flow: %s", GST_OBJECT_NAME(appsrc), gst_flow_get_name(ret)));
- gst_task_stop(src->priv->task.get());
- }
- } else
- gst_buffer_unref(channelBuffer);
+ auto& appsrc = priv->sources[i];
+ // Leak the buffer ref, because gst_app_src_push_buffer steals it.
+ GstFlowReturn ret = gst_app_src_push_buffer(GST_APP_SRC(appsrc.get()), buffer.leakRef());
+ if (ret != GST_FLOW_OK) {
+ // FLUSHING and EOS are not errors.
+ if (ret < GST_FLOW_EOS || ret == GST_FLOW_NOT_LINKED)
+ GST_ELEMENT_ERROR(src, CORE, PAD, ("Internal WebAudioSrc error"), ("Failed to push buffer on %s flow: %s", GST_OBJECT_NAME(appsrc.get()), gst_flow_get_name(ret)));
+ gst_task_stop(src->priv->task.get());
+ failed = true;
+ }
}
-
- g_slist_free(channelBufferList);
}
static GstStateChangeReturn webKitWebAudioSrcChangeState(GstElement* element, GstStateChange transition)
@@ -414,11 +396,12 @@
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED: {
GST_DEBUG_OBJECT(src, "READY->PAUSED");
+
src->priv->pool = gst_buffer_pool_new();
- GstStructure* config = gst_buffer_pool_get_config(src->priv->pool);
+ GstStructure* config = gst_buffer_pool_get_config(src->priv->pool.get());
gst_buffer_pool_config_set_params(config, nullptr, src->priv->bufferSize, 0, 0);
- gst_buffer_pool_set_config(src->priv->pool, config);
- if (!gst_buffer_pool_set_active(src->priv->pool, TRUE))
+ gst_buffer_pool_set_config(src->priv->pool.get(), config);
+ if (!gst_buffer_pool_set_active(src->priv->pool.get(), TRUE))
returnValue = GST_STATE_CHANGE_FAILURE;
else if (!gst_task_start(src->priv->task.get()))
returnValue = GST_STATE_CHANGE_FAILURE;
@@ -426,13 +409,13 @@
}
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT(src, "PAUSED->READY");
+
#if GST_CHECK_VERSION(1, 4, 0)
- gst_buffer_pool_set_flushing(src->priv->pool, TRUE);
+ gst_buffer_pool_set_flushing(src->priv->pool.get(), TRUE);
#endif
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);
+ gst_buffer_pool_set_active(src->priv->pool.get(), FALSE);
src->priv->pool = nullptr;
break;
default:
Modified: trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerUtilities.cpp (210583 => 210584)
--- trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerUtilities.cpp 2017-01-11 09:08:11 UTC (rev 210583)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/GStreamerUtilities.cpp 2017-01-11 11:27:38 UTC (rev 210584)
@@ -120,10 +120,10 @@
return reinterpret_cast<char*>(mapInfo->data);
}
-void mapGstBuffer(GstBuffer* buffer)
+void mapGstBuffer(GstBuffer* buffer, uint32_t flags)
{
GstMapInfo* mapInfo = static_cast<GstMapInfo*>(fastMalloc(sizeof(GstMapInfo)));
- if (!gst_buffer_map(buffer, mapInfo, GST_MAP_WRITE)) {
+ if (!gst_buffer_map(buffer, mapInfo, static_cast<GstMapFlags>(flags))) {
fastFree(mapInfo);
gst_buffer_unref(buffer);
return;
@@ -130,7 +130,7 @@
}
GstMiniObject* miniObject = reinterpret_cast<GstMiniObject*>(buffer);
- gst_mini_object_set_qdata(miniObject, g_quark_from_static_string(webkitGstMapInfoQuarkString), mapInfo, 0);
+ gst_mini_object_set_qdata(miniObject, g_quark_from_static_string(webkitGstMapInfoQuarkString), mapInfo, nullptr);
}
void unmapGstBuffer(GstBuffer* buffer)
